Re: [asterisk-users] ODBC Connection Problem
First of all test your odbc-connection via console: isql telco-ops dba c3podb@2012 -v You should see a Connected!-Message. Do you? Second: yes I also had problems setting up odbc. The main problem/error for me was, that documentation is sometimes confusing. Here is my config. Please notice the [section] - namings: /etc/odbcinst.ini [MySQL] Description = MySQL ODBCMyODBC Driver Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so FileUsage = 1 /etc/odbc.ini [MySQL-asterisk] Description = MySQL ODBC Driver Driver = MySQL Socket = /var/run/mysqld/mysqld.sock Server = localhost User = my_username Password = my_password Database = my_database Option = 3 Port = Charset = utf8 /etc/asterisk/res_odbc.conf [mysql] enabled = yes dsn = MySQL-asterisk username = my_username password = my_password pre-connect = yes /etc/asterisk/cdr_odbc.conf [global] dsn=mysql loguniqueid=yes dispositionstring=yes table=cdr /etc/asterisk/cel_odbc.conf [first] connection=mysql table=cel| | Additionally you will need some configurations for you realtime-config. This config above is only for cdr- and cel-logging via odbc. -Thorsten- Am 10.12.2012 12:23, schrieb Chandrakant Solanki: /etc/odbc.ini [telco-ops] Description = Asterisk realtime and other FUNC_ODBC access Driver = MySQL Server = 172.18.100.18 Socket = /var/lib/mysql/data3306/mysql.sock User= dba Password= c3podb@2012 Database= mytelcoexample Port= 3306 Option = 3 On Mon, Dec 10, 2012 at 4:34 PM, Thorsten Göllner t...@ovm-group.com mailto:t...@ovm-group.com wrote: Am 10.12.2012 06:37, schrieb Chandrakant Solanki: Hi All, OS : CentOS 5 64bit OS Machine Asterisk: 1.8.13.0 ODBC Packages: unixODBC-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2 unixODBC-devel-2.2.11-7.1 res_odbc.conf [telco-ops] enabled = yes dsn = telco-ops username = dba password = c3podb@2012 pre-connect = yes sanitysql = select 1 idlecheck = 15 ;isolation = repeatable_read pooling = yes limit = 3600 connect_timeout = 10 negative_connection_cache = 30 Above is my installation package and configuration file (res_odbc.conf), when I try to execute odbc show all it always gives below output. *CLI odbc show all ODBC DSN Settings - Name: telco-ops DSN:telco-ops Last connection attempt: 1970-01-01 00:00:00 Pooled: Yes Limit: 3600 Connections in use: 1 - Connection 1: connected When Insert/Update/Select query will be executed, it can't update last connection attempt field. In result, ODBC stuck after few minutes, and in this case I also need to restart asterisk, because I can't type any command, it can't give any command's output. Also updated asterisk with 10.9.0, but same result. Please show us /etc/odbc.ini too. -- _ -- 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] date - outgoing call
On Tuesday 11 December 2012, Joseph wrote:
On 12/10/12 20:45, Steve Edwards wrote:
On Mon, 10 Dec 2012, Joseph wrote:
When a call comes in asterisk records the date correctly but when I cake
a call out I get only something like:
Date: 60
here is an example:
From: 7807560785
To:s
Date: 2012-12-11 00:46:04
Status:ANSWERED
From: 5
To:4331235
Date: 60
Status:4
What version of Asterisk?
Where are you seeing the data you display above?
I'm using this command to pull the records of the cvs:
grep -P '\d{4}-\d{2}-\d{2}\s\d{2}:\d{2}:\d{2}'
/var/log/asterisk/cdr-csv/*.csv | cut -d, -f2,3,10,15 | awk -F, {'print
From:\t$1\nTo:\t$2\nDate:\t$3\nStatus:\t$4\n\n'}
|tail -60
the actual last records in Master.csv is:
,5,218,internal,Cerra
5,IAX2/192.168.141.1:4569-4374,SIP/11-0180,Dial,SIP/11SIP/3
21SIP/218,25,m(penguin)w,2012-12-11 02:58:53,2012-12-11
02:58:59,2012-12-11
03:00:00,67,61,ANSWERED,DOCUMENTATION,1355194733.524,
but with the above command it display:
From: 5
To: 218
Date: 25
Status: 67
Your problem is, the parameter string to Dial contains embedded commas
(although they are in between speech marks, and therefore perfectly valid in
the context of a CSV file). Your naïve split on commas is failing to account
for these. So SIP/11SIP/321SIP/218,25,m(penguin)w appears to be three
separate fields: \SIP/11SIP/321SIP/218 , 25 and m(penguin)w\ -- and
this is what is throwing your simple script out.
You almost certainly will need to use a better scripting language and proper
CSV parsing library. In Perl, there is Text::CSV, which is available from
CPAN or (in Debian / Ubuntu) using
$ sudo apt-get install libtext-csv-perl
and there must be similar things in other scripting languages.
Alternatively, you could use MySQL for CDR, and SELECT only the fields you want
-- but for licensing reasons you will have to compile the MySQL extension
yourself, from Source Code.
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] date - outgoing call
On Mon, Dec 10, 2012 at 11:02 PM, Joseph syscon...@gmail.com wrote:
On 12/10/12 20:45, Steve Edwards wrote:
On Mon, 10 Dec 2012, Joseph wrote:
When a call comes in asterisk records the date correctly but when I cake
a
call out I get only something like:
Date: 60
here is an example:
From: 7807560785
To: s
Date: 2012-12-11 00:46:04
Status: ANSWERED
From: 5
To: 4331235
Date: 60
Status: 4
What version of Asterisk?
Where are you seeing the data you display above?
--
Thanks in advance,
--**--**
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
I'm using this command to pull the records of the cvs:
grep -P '\d{4}-\d{2}-\d{2}\s\d{2}:\d{**2}:\d{2}'
/var/log/asterisk/cdr-csv/*.**csv | cut -d, -f2,3,10,15 | awk -F,
{'print
From:\t$1\nTo:\t$2\nDate:**\t$3\nStatus:\t$4\n---**-\n'}
|tail -60
the actual last records in Master.csv is:
,5,218,internal,**Cerra 5,IAX2/192.168.141.1:4569-**
4374,SIP/11-0180,Dial**,SIP/11SIP/321SIP/218,25,m(**penguin)w,2012-12-11
02:58:53,2012-12-11 02:58:59,2012-12-11 03:00:00,67,61,ANSWERED,**
DOCUMENTATION,1355194733.**524,
but with the above command it display:
From: 5
To: 218
Date: 25
Status: 67
--
Joseph
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The problem is not exactly Asterisk, it is how the data being parsed from
the CSV file. If you look at the data, each field is enclosed with double
quotes. One field in particular may have commas embedded within the double
quotes. For instance, in your data there is one field that looks like this:
SIP/11SIP/321SIP/218,25,m(penguin)w
Using the 'cut' utility does not take into account the quoiting, so cut
happily chops that one field into three. This causes the date and status
field numbers to be different (according to cut) and therefore your request
for fields 10 and 15 return the wrong columns. To compound the issue, some
records do not have the extra commas so you cannot simply change the field
numbers.
I have not spent any real time looking at a way to deal with this using
command line utilities because I read the data from a mysql database.
Another option would be to read the data into a spreadsheet. You could
also write a script around what you are doing and parts records differently
based on the 'last appliation' such as Dial, Read, Voicemail.
I hope that helps.
Dale
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Re: [asterisk-users] date - outgoing call
On Tuesday 11 December 2012, Dale Noll wrote: . [stuff deleted] . You could also write a script around what you are doing and parts records differently based on the 'last appliation' such as Dial, Read, Voicemail. Prioblem: You still don't know how many commas are going to be present in the application data field. You really need a proper CSV parsing library. It's fun to write one for a programming exercise, but easier to use an existing Open Source one if possible. -- 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] callerid not received from dahdi
On Tue, Dec 11, 2012 at 10:07:42AM +0530, Harish Mandowara wrote: Hi, Thank you for your reply. 77 ext. number is connected with my asterisk. so any one want to talk with jitsi(pc), they have to dial 77 then 2000#(jitsi sip user number). my pbx is sending callerid. i can see on other analog phone display. Yes pbx is sending callerid. When i dial any ext. number from jitsi. On the recipient phone display shows 77 ext number. i tried all combination from https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India but it does not work. Does the manual for your PBX say anything about how it's sending the CallerID information? Knowing specifically how the CallerID is supposed to be sent can allow us to more quickly narrow in on why you may not be detecting it. Also, in the configs you originally posted you had #include dahdi-channels.conf. Would it be possible to send that configuration options you have attached in that file as well? Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
[asterisk-users] [asterisk] Guide for setup a server for end2end video call
Dear List, Where can I find a guide for setup an Asterisk server which can eastanblish a simple video call from two sip clients? Thank you! Regards, Barco -- _ -- 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] disconnect supervision
Most of the time my phone line are working OK but at time to time when I run: asterisk -rx core show channels it show: Channel Location State Application(Data) SIP/pstn--00 (None) Up AppDial((Outgoing Line)) SIP/pstn-9998-00 7807586576@internal: Up Dial(SIP/97807807586576@pstn-4 2 active channels 1 active call even though nobody is using any line. I'm using Audiocodes gateway. Does it have anything to do with disconnect supervision on analog line in Audiocodes gateway? -- Joseph -- _ -- 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] disconnect supervision
Could be, but I'd check the easier to fix polarity settings. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 11, 2012 11:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] disconnect supervision Most of the time my phone line are working OK but at time to time when I run: asterisk -rx core show channels it show: Channel Location State Application(Data) SIP/pstn--00 (None) Up AppDial((Outgoing Line)) SIP/pstn-9998-00 7807586576@internal: Up Dial(SIP/97807807586576@pstn-4 2 active channels 1 active call even though nobody is using any line. I'm using Audiocodes gateway. Does it have anything to do with disconnect supervision on analog line in Audiocodes gateway? -- Joseph -- _ -- 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] disconnect supervision
On 12/11/12 11:30, Danny Nicholas wrote: Could be, but I'd check the easier to fix polarity settings. How do I do that? Notice, that this channel hang-up/disconnect does not happen all the time, only once a while could be once a day or once a week. -- Joseph -- _ -- 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] disconnect supervision
In /etc/asterisk/dahdi.conf, check your answeronpolarityswitch and hanguponpolarityswitch lines. If they aren't present, the default values are being used. If they are, tweak them and restart asterisk and dahdi. I do this - service asterisk stop; service dahdi restart; service asterisk start. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 11, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] disconnect supervision On 12/11/12 11:30, Danny Nicholas wrote: Could be, but I'd check the easier to fix polarity settings. How do I do that? Notice, that this channel hang-up/disconnect does not happen all the time, only once a while could be once a day or once a week. -- Joseph -- _ -- 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] disconnect supervision
On 12/11/12 11:48, Danny Nicholas wrote: In /etc/asterisk/dahdi.conf, check your answeronpolarityswitch and hanguponpolarityswitch lines. If they aren't present, the default values are being used. If they are, tweak them and restart asterisk and dahdi. I do this - service asterisk stop; service dahdi restart; service asterisk start. I'm not using dahdi.conf I'm using extension.conf sip.conf with analog AudioCodes gateway FXO/FXS -- Joseph -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 11, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] disconnect supervision On 12/11/12 11:30, Danny Nicholas wrote: Could be, but I'd check the easier to fix polarity settings. How do I do that? Notice, that this channel hang-up/disconnect does not happen all the time, only once a while could be once a day or once a week. -- Joseph -- _ -- 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] disconnect supervision
You need to look at the device which the analog lines plug into. There is nothing to change in Asterisk for this issue. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 11, 2012 12:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] disconnect supervision On 12/11/12 11:48, Danny Nicholas wrote: In /etc/asterisk/dahdi.conf, check your answeronpolarityswitch and hanguponpolarityswitch lines. If they aren't present, the default values are being used. If they are, tweak them and restart asterisk and dahdi. I do this - service asterisk stop; service dahdi restart; service asterisk start. I'm not using dahdi.conf I'm using extension.conf sip.conf with analog AudioCodes gateway FXO/FXS -- Joseph -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 11, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] disconnect supervision On 12/11/12 11:30, Danny Nicholas wrote: Could be, but I'd check the easier to fix polarity settings. How do I do that? Notice, that this channel hang-up/disconnect does not happen all the time, only once a while could be once a day or once a week. -- Joseph -- _ -- 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] disconnect supervision
If you are using an analogue/sip ata, then the problem is on the ata. Run a packet capture and you'll see the invite coming from the ata without nobody using the phone... I am typing from my mobile phone... Il giorno 11/dic/2012 18:55, Joseph syscon...@gmail.com ha scritto: On 12/11/12 11:48, Danny Nicholas wrote: In /etc/asterisk/dahdi.conf, check your answeronpolarityswitch and hanguponpolarityswitch lines. If they aren't present, the default values are being used. If they are, tweak them and restart asterisk and dahdi. I do this - service asterisk stop; service dahdi restart; service asterisk start. I'm not using dahdi.conf I'm using extension.conf sip.conf with analog AudioCodes gateway FXO/FXS -- Joseph -Original Message- From: asterisk-users-bounces@lists.**digium.comasterisk-users-boun...@lists.digium.com [mailto:asterisk-users-**boun...@lists.digium.comasterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 11, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] disconnect supervision On 12/11/12 11:30, Danny Nicholas wrote: Could be, but I'd check the easier to fix polarity settings. How do I do that? Notice, that this channel hang-up/disconnect does not happen all the time, only once a while could be once a day or once a week. -- Joseph -- __**__**_ -- 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
[asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715
I have an asterisk server at home. I'm looking to replace my internal phones with sip cordless (DECT) phones. I'm now looking at the Siemens A510IP base ($90 ) and A510H handset ($40) and the Grandview DP715 base ($80) and DP710 handset ($45). The Siemens has a feature were I can also use a PSTN landline, but I not sure that's a great benefit. Has anybody tried these phones? I assume they both integrate with asterisk since they both are sip. I'm leaning towards Grandview. Seems to be easier to transfer calls. A questions, I'm on a call with one handset, can another person pick up a second handset to make another call? can that person join the first call? In other words, do I need two base stations to make two calls? Any suggestions? Thoughts? sean -- _ -- 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] DECT phone for home: siemens A510 v. Grandstream DP715
I've been using the Gigaset A580 Base and A58H Phone for about 3 years now. Never gave me problems. The call Quality is excellent! I only have 1 handset connected to the Base but I want more. I bought a Linksys WIP330 as a 2nd phone to try out and that works just as good without a base unit. The A580 Base supports up to 6 handsets. I have 6 Incoming VOIP Numbers using separate SIP accounts pointed to 1 Handset but you can point each SIP to separate handsets. The call goes to the first phone that picks up. When on a call, picking up another phone makes a separate call and does not conference. I don't use conference yet but I know you have to put the call on hold or something. The thing I don't like about the A580 and might be the same on all of them is that you can only specify 1 Sip Account for making outgoing calls. In other words, all 6 phones would use the same caller id out, but I wanted to be able to choose that because I have a business number and number for each person in our household. In order to use a different Caller ID (SIP Account) for making outgoing calls I added a extension to my Dial Plan and before making outgoing calls I press *1-6 before the number. I'm going to try adding more handsets that are compatible. I want the SL78H but they are so expensive for just home everyday use. Make sure you check the compatibility page here before buying handsets. http://gigaset.com/us/en/cms/PageCustomerServicesCompatibility.html Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 11:32 AM, sean darcy wrote: Siemens A510IP -- _ -- 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] Guide for setup a server for end2end video call
On Tue, 2012-12-11 at 23:02 +0800, Barco You wrote: Dear List, Where can I find a guide for setup an Asterisk server which can eastanblish a simple video call from two sip clients? Thank you! Regards, Barco Hi Barco, I don't think there is a specific guide for this. From the top of my head.. In /etc/asterisk/sip.conf, you will find the the default setting is _not_ to have video enbled. So either you enable it gloabally (for all sip-users) or individually for specific sip-users. Greatest pit-fall. you have to analyse the codecs for all hard-phones and/or softphones. You should have atleast one common codec enabled. If not, you will only get an audio connection (unfortunately, without any warning) Most safe option (atleast to begin with) is to use same clients at both sides, and configure them identically. Second suggestion, is to define a echo-function. If you dial the defined extension, you get not only audio echo, but also video-echo. I've been testing with a couple of clients, with various results. hans -- _ -- 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] DECT phone for home: siemens A510 v. Grandstream DP715
One thing I dislike about the A580H is that the handset always says 'You have new messages' if I've missed a call. It wouldn't bug me if it said 'missed call' but it tells me I have new messages and even lights up a red LED under a button with a picture of an envelope on it. I'm about to test an A510IP and an A610IP to compare against the A580. Fingers crossed neither of them has that issue, because the Gigaset phone is a pretty good phone other than that, and the difficulty doing a (blind) transfer, as referred to by the OP. Pete On 12/12/2012, at 8:57 AM, Roy Abshire r...@coopvr.com wrote: I've been using the Gigaset A580 Base and A58H Phone for about 3 years now. Never gave me problems. The call Quality is excellent! I only have 1 handset connected to the Base but I want more. I bought a Linksys WIP330 as a 2nd phone to try out and that works just as good without a base unit. The A580 Base supports up to 6 handsets. I have 6 Incoming VOIP Numbers using separate SIP accounts pointed to 1 Handset but you can point each SIP to separate handsets. The call goes to the first phone that picks up. When on a call, picking up another phone makes a separate call and does not conference. I don't use conference yet but I know you have to put the call on hold or something. The thing I don't like about the A580 and might be the same on all of them is that you can only specify 1 Sip Account for making outgoing calls. In other words, all 6 phones would use the same caller id out, but I wanted to be able to choose that because I have a business number and number for each person in our household. In order to use a different Caller ID (SIP Account) for making outgoing calls I added a extension to my Dial Plan and before making outgoing calls I press *1-6 before the number. I'm going to try adding more handsets that are compatible. I want the SL78H but they are so expensive for just home everyday use. Make sure you check the compatibility page here before buying handsets. http://gigaset.com/us/en/cms/PageCustomerServicesCompatibility.html Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 11:32 AM, sean darcy wrote: Siemens A510IP smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] DECT phone for home: siemens A510 v. Grandstream DP715
That is true about the A580. I don't like the interface much to check messages. Besides that every time I go to dial a number...it always uses the first digit pressed to go into phone mode..so I have to press the first digit twice... I would test other phones but it's for home and I can't fork over $$ to try them all out I have tested some Nokia cell phones, the N97, N900, and E71 and the E71 and N900 worked well. I didn't like the N97. Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 12:52 PM, Pete Mundy wrote: One thing I dislike about the A580H is that the handset always says 'You have new messages' if I've missed a call. It wouldn't bug me if it said 'missed call' but it tells me I have new messages and even lights up a red LED under a button with a picture of an envelope on it. I'm about to test an A510IP and an A610IP to compare against the A580. Fingers crossed neither of them has that issue, because the Gigaset phone is a pretty good phone other than that, and the difficulty doing a (blind) transfer, as referred to by the OP. Pete On 12/12/2012, at 8:57 AM, Roy Abshire r...@coopvr.com wrote: I've been using the Gigaset A580 Base and A58H Phone for about 3 years now. Never gave me problems. The call Quality is excellent! I only have 1 handset connected to the Base but I want more. I bought a Linksys WIP330 as a 2nd phone to try out and that works just as good without a base unit. The A580 Base supports up to 6 handsets. I have 6 Incoming VOIP Numbers using separate SIP accounts pointed to 1 Handset but you can point each SIP to separate handsets. The call goes to the first phone that picks up. When on a call, picking up another phone makes a separate call and does not conference. I don't use conference yet but I know you have to put the call on hold or something. The thing I don't like about the A580 and might be the same on all of them is that you can only specify 1 Sip Account for making outgoing calls. In other words, all 6 phones would use the same caller id out, but I wanted to be able to choose that because I have a business number and number for each person in our household. In order to use a different Caller ID (SIP Account) for making outgoing calls I added a extension to my Dial Plan and before making outgoing calls I press *1-6 before the number. I'm going to try adding more handsets that are compatible. I want the SL78H but they are so expensive for just home everyday use. Make sure you check the compatibility page here before buying handsets. http://gigaset.com/us/en/cms/PageCustomerServicesCompatibility.html Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 11:32 AM, sean darcy wrote: Siemens A510IP -- _ -- 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] MACRO_CONTEXT equivalent for GoSub
Is there an equivalent of MACRO_CONTEXT for a GoSub? Looking for a way to determine the name of the calling context. -- Mitch -- _ -- 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] MACRO_CONTEXT equivalent for GoSub
You don't state version, but I'm pretty sure this animal doesn't exist.
What I did in 1.4 was to set a variable before the gosub so I could track
it. Something like this
Exten = s,n,Set(from=foo)
Exten = s,n,gosub(showfoo,s,1)
Exten = s,n,Set(from=bar)
Exten = s,n,gosub(showfoo,s,1)
[showfoo]
Exten = s,1,verbose(called from ${from})
Exten = s,n,return()
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn
Sent: Tuesday, December 11, 2012 3:52 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MACRO_CONTEXT equivalent for GoSub
Is there an equivalent of MACRO_CONTEXT for a GoSub? Looking for a way
to determine the name of the calling context.
--
Mitch
--
_
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Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715
On 12/11/2012 04:37 PM, Roy Abshire wrote: That is true about the A580. I don't like the interface much to check messages. Besides that every time I go to dial a number...it always uses the first digit pressed to go into phone mode..so I have to press the first digit twice... I would test other phones but it's for home and I can't fork over $$ to try them all out I have tested some Nokia cell phones, the N97, N900, and E71 and the E71 and N900 worked well. I didn't like the N97. Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 12:52 PM, Pete Mundy wrote: One thing I dislike about the A580H is that the handset always says 'You have new messages' if I've missed a call. It wouldn't bug me if it said 'missed call' but it tells me I have new messages and even lights up a red LED under a button with a picture of an envelope on it. I'm about to test an A510IP and an A610IP to compare against the A580. Fingers crossed neither of them has that issue, because the Gigaset phone is a pretty good phone other than that, and the difficulty doing a (blind) transfer, as referred to by the OP. Pete On 12/12/2012, at 8:57 AM, Roy Abshirer...@coopvr.com wrote: I've been using the Gigaset A580 Base and A58H Phone for about 3 years now. Never gave me problems. The call Quality is excellent! I only have 1 handset connected to the Base but I want more. I bought a Linksys WIP330 as a 2nd phone to try out and that works just as good without a base unit. The A580 Base supports up to 6 handsets. I have 6 Incoming VOIP Numbers using separate SIP accounts pointed to 1 Handset but you can point each SIP to separate handsets. The call goes to the first phone that picks up. When on a call, picking up another phone makes a separate call and does not conference. I don't use conference yet but I know you have to put the call on hold or something. The thing I don't like about the A580 and might be the same on all of them is that you can only specify 1 Sip Account for making outgoing calls. In other words, all 6 phones would use the same caller id out, but I wanted to be able to choose that because I have a business number and number for each person in our household. In order to use a different Caller ID (SIP Account) for making outgoing calls I added a extension to my Dial Plan and before making outgoing calls I press *1-6 before the number. I'm going to try adding more handsets that are compatible. I want the SL78H but they are so expensive for just home everyday use. Make sure you check the compatibility page here before buying handsets. http://gigaset.com/us/en/cms/PageCustomerServicesCompatibility.html Some stupid questions: I understand the A510 allows 2 sip calls. Let's say you've registered the base with asterisk. A uses handset 1 to call out over sip. B picks up handset 2. Does B hear a dial tone? Can B dial out over the asterisk server? Or do you need two registrations with asterisk? In which case, is handset 2 always tied to the second registration? sean -- _ -- 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] MACRO_CONTEXT equivalent for GoSub
Was looking for 1.8 and above. I ended up doing something similar to
what you describe. Not terribly elegant, but it works.
Mitch
On 12/11/2012 04:03 PM, Danny Nicholas wrote:
You don't state version, but I'm pretty sure this animal doesn't exist.
What I did in 1.4 was to set a variable before the gosub so I could track
it. Something like this
Exten = s,n,Set(from=foo)
Exten = s,n,gosub(showfoo,s,1)
Exten = s,n,Set(from=bar)
Exten = s,n,gosub(showfoo,s,1)
[showfoo]
Exten = s,1,verbose(called from ${from})
Exten = s,n,return()
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn
Sent: Tuesday, December 11, 2012 3:52 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MACRO_CONTEXT equivalent for GoSub
Is there an equivalent of MACRO_CONTEXT for a GoSub? Looking for a way
to determine the name of the calling context.
--
_
-- 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] DECT phone for home: siemens A510 v. Grandstream DP715
I never hear a dial tone from my sip phones. I just pick up the phone and dial + send but you should be able to dial out using the same SIP account in use...but you will need at least 2 outgoing trunks with your SIP provider to call external numbers, unless your calling another extension. Just picking up your phone will not connect you to the call in progress on the other line. Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 2:39 PM, sean darcy wrote: On 12/11/2012 04:37 PM, Roy Abshire wrote: That is true about the A580. I don't like the interface much to check messages. Besides that every time I go to dial a number...it always uses the first digit pressed to go into phone mode..so I have to press the first digit twice... I would test other phones but it's for home and I can't fork over $$ to try them all out I have tested some Nokia cell phones, the N97, N900, and E71 and the E71 and N900 worked well. I didn't like the N97. Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 12:52 PM, Pete Mundy wrote: One thing I dislike about the A580H is that the handset always says 'You have new messages' if I've missed a call. It wouldn't bug me if it said 'missed call' but it tells me I have new messages and even lights up a red LED under a button with a picture of an envelope on it. I'm about to test an A510IP and an A610IP to compare against the A580. Fingers crossed neither of them has that issue, because the Gigaset phone is a pretty good phone other than that, and the difficulty doing a (blind) transfer, as referred to by the OP. Pete On 12/12/2012, at 8:57 AM, Roy Abshirer...@coopvr.com wrote: I've been using the Gigaset A580 Base and A58H Phone for about 3 years now. Never gave me problems. The call Quality is excellent! I only have 1 handset connected to the Base but I want more. I bought a Linksys WIP330 as a 2nd phone to try out and that works just as good without a base unit. The A580 Base supports up to 6 handsets. I have 6 Incoming VOIP Numbers using separate SIP accounts pointed to 1 Handset but you can point each SIP to separate handsets. The call goes to the first phone that picks up. When on a call, picking up another phone makes a separate call and does not conference. I don't use conference yet but I know you have to put the call on hold or something. The thing I don't like about the A580 and might be the same on all of them is that you can only specify 1 Sip Account for making outgoing calls. In other words, all 6 phones would use the same caller id out, but I wanted to be able to choose that because I have a business number and number for each person in our household. In order to use a different Caller ID (SIP Account) for making outgoing calls I added a extension to my Dial Plan and before making outgoing calls I press *1-6 before the number. I'm going to try adding more handsets that are compatible. I want the SL78H but they are so expensive for just home everyday use. Make sure you check the compatibility page here before buying handsets. http://gigaset.com/us/en/cms/PageCustomerServicesCompatibility.html Some stupid questions: I understand the A510 allows 2 sip calls. Let's say you've registered the base with asterisk. A uses handset 1 to call out over sip. B picks up handset 2. Does B hear a dial tone? Can B dial out over the asterisk server? Or do you need two registrations with asterisk? In which case, is handset 2 always tied to the second registration? sean -- _ -- 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] DECT phone for home: siemens A510 v. Grandstream DP715
Using a Gigaset C610IP here, and am very happy with the features. The base station can handle two concurrent SIP calls, and another internal one at that. It does it with a single SIP registration to each server. You can setup multiple servers if you want to and define dial patterns/plans that determine which server gets used. After some playing around with it, I'm now using my setup connected to a single asterisk only. (Let asterisk make call routing decisions based on cost, using an AGI) Call transfer is working fine, the handsets have a Flash/R key to accomplish this. Using the Flash lets you start a second call, and once answered you can easily conference the second party in (softkey right on the screen), or transfer the call to the other party (via menu, then transfer). Using this capability, someone on a call can easily confer with another party, and bridge them into the call. AFAIK it is not possible for someone to join an existing call easily. You'd have to implement that in asterisk's dialplan, not on the Gigaset phone. My understanding is that the C610IP has a few more features than the 510. I might've also read somewhere that the 510 is obsolete. Can't find that link right now, but search mgraves.org (use the Gigaset tag to get some initial results). On Tue, Dec 11, 2012 at 2:37 PM, Roy Abshire r...@coopvr.com wrote: That is true about the A580. I don't like the interface much to check messages. Besides that every time I go to dial a number...it always uses the first digit pressed to go into phone mode..so I have to press the first digit twice... I would test other phones but it's for home and I can't fork over $$ to try them all out I have tested some Nokia cell phones, the N97, N900, and E71 and the E71 and N900 worked well. I didn't like the N97. Co-op Vacation Rentalswww.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 12:52 PM, Pete Mundy wrote: One thing I dislike about the A580H is that the handset always says 'You have new messages' if I've missed a call. It wouldn't bug me if it said 'missed call' but it tells me I have new messages and even lights up a red LED under a button with a picture of an envelope on it. I'm about to test an A510IP and an A610IP to compare against the A580. Fingers crossed neither of them has that issue, because the Gigaset phone is a pretty good phone other than that, and the difficulty doing a (blind) transfer, as referred to by the OP. Pete On 12/12/2012, at 8:57 AM, Roy Abshire r...@coopvr.com r...@coopvr.com wrote: I've been using the Gigaset A580 Base and A58H Phone for about 3 years now. Never gave me problems. The call Quality is excellent! I only have 1 handset connected to the Base but I want more. I bought a Linksys WIP330 as a 2nd phone to try out and that works just as good without a base unit. The A580 Base supports up to 6 handsets. I have 6 Incoming VOIP Numbers using separate SIP accounts pointed to 1 Handset but you can point each SIP to separate handsets. The call goes to the first phone that picks up. When on a call, picking up another phone makes a separate call and does not conference. I don't use conference yet but I know you have to put the call on hold or something. The thing I don't like about the A580 and might be the same on all of them is that you can only specify 1 Sip Account for making outgoing calls. In other words, all 6 phones would use the same caller id out, but I wanted to be able to choose that because I have a business number and number for each person in our household. In order to use a different Caller ID (SIP Account) for making outgoing calls I added a extension to my Dial Plan and before making outgoing calls I press *1-6 before the number. I'm going to try adding more handsets that are compatible. I want the SL78H but they are so expensive for just home everyday use. Make sure you check the compatibility page here before buying handsets. http://gigaset.com/us/en/cms/PageCustomerServicesCompatibility.html Co-op Vacation Rentalswww.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 11:32 AM, sean darcy wrote: Siemens A510IP -- _ -- 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
Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715
Mebbe you guys should try snom m9 dect ip phone, i have been using it since over 3 years now without any of these issues. Mitul On Dec 12, 2012 4:25 AM, Kai-Uwe Jensen kujen...@gmail.com wrote: Using a Gigaset C610IP here, and am very happy with the features. The base station can handle two concurrent SIP calls, and another internal one at that. It does it with a single SIP registration to each server. You can setup multiple servers if you want to and define dial patterns/plans that determine which server gets used. After some playing around with it, I'm now using my setup connected to a single asterisk only. (Let asterisk make call routing decisions based on cost, using an AGI) Call transfer is working fine, the handsets have a Flash/R key to accomplish this. Using the Flash lets you start a second call, and once answered you can easily conference the second party in (softkey right on the screen), or transfer the call to the other party (via menu, then transfer). Using this capability, someone on a call can easily confer with another party, and bridge them into the call. AFAIK it is not possible for someone to join an existing call easily. You'd have to implement that in asterisk's dialplan, not on the Gigaset phone. My understanding is that the C610IP has a few more features than the 510. I might've also read somewhere that the 510 is obsolete. Can't find that link right now, but search mgraves.org (use the Gigaset tag to get some initial results). On Tue, Dec 11, 2012 at 2:37 PM, Roy Abshire r...@coopvr.com wrote: That is true about the A580. I don't like the interface much to check messages. Besides that every time I go to dial a number...it always uses the first digit pressed to go into phone mode..so I have to press the first digit twice... I would test other phones but it's for home and I can't fork over $$ to try them all out I have tested some Nokia cell phones, the N97, N900, and E71 and the E71 and N900 worked well. I didn't like the N97. Co-op Vacation Rentalswww.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 12:52 PM, Pete Mundy wrote: One thing I dislike about the A580H is that the handset always says 'You have new messages' if I've missed a call. It wouldn't bug me if it said 'missed call' but it tells me I have new messages and even lights up a red LED under a button with a picture of an envelope on it. I'm about to test an A510IP and an A610IP to compare against the A580. Fingers crossed neither of them has that issue, because the Gigaset phone is a pretty good phone other than that, and the difficulty doing a (blind) transfer, as referred to by the OP. Pete On 12/12/2012, at 8:57 AM, Roy Abshire r...@coopvr.com r...@coopvr.com wrote: I've been using the Gigaset A580 Base and A58H Phone for about 3 years now. Never gave me problems. The call Quality is excellent! I only have 1 handset connected to the Base but I want more. I bought a Linksys WIP330 as a 2nd phone to try out and that works just as good without a base unit. The A580 Base supports up to 6 handsets. I have 6 Incoming VOIP Numbers using separate SIP accounts pointed to 1 Handset but you can point each SIP to separate handsets. The call goes to the first phone that picks up. When on a call, picking up another phone makes a separate call and does not conference. I don't use conference yet but I know you have to put the call on hold or something. The thing I don't like about the A580 and might be the same on all of them is that you can only specify 1 Sip Account for making outgoing calls. In other words, all 6 phones would use the same caller id out, but I wanted to be able to choose that because I have a business number and number for each person in our household. In order to use a different Caller ID (SIP Account) for making outgoing calls I added a extension to my Dial Plan and before making outgoing calls I press *1-6 before the number. I'm going to try adding more handsets that are compatible. I want the SL78H but they are so expensive for just home everyday use. Make sure you check the compatibility page here before buying handsets. http://gigaset.com/us/en/cms/PageCustomerServicesCompatibility.html Co-op Vacation Rentalswww.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 11:32 AM, sean darcy wrote: Siemens A510IP -- _ -- 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] DECT phone for home: siemens A510 v. Grandstream DP715
At the risk of continuing off-topic conversation... Oh the M9 has it's own issues, don't you worry (not to mention it's _way_ more expensive than the Gigaset range)! I've been testing the A510 today and I've decided I like it more than the A580. The software (via the web interface) looks more polished and overall thus far it has performed fairly well (with no annoying 'new message' statement!). I haven't tried multiple SIP registrations with dual handsets yet, but I intend to. You guys realise these are standardised DECT handsets right? Ie that's where the 6-handset limitation comes from, and you should theoretically be able to register other manufacturers' handsets too. I'm about to try the C610IP model that Kai-Uwe mentioned (over the next few days). I think I'll like even more than the A580 since it has a higher resolution colour display. And even then it's still miles cheaper than the M9! Don't get me wrong, I do really like Snom phones too, although the M9 hasn't been entirely smooth sailing. I love the 320 desk phone (at the cheaper end of the scale, it has a great feel to it). Isn't it great that they all interact with Asterisk so well (there you go, I'm on-topic again ;-). Pete On 12/12/2012, at 4:12 PM, Mitul Limbani mi...@enterux.in wrote: Mebbe you guys should try snom m9 dect ip phone, i have been using it since over 3 years now without any of these issues. Mitul On Dec 12, 2012 4:25 AM, Kai-Uwe Jensen kujen...@gmail.com wrote: Using a Gigaset C610IP here, and am very happy with the features. The base station can handle two concurrent SIP calls, and another internal one at that. It does it with a single SIP registration to each server. You can setup multiple servers if you want to and define dial patterns/plans that determine which server gets used. After some playing around with it, I'm now using my setup connected to a single asterisk only. (Let asterisk make call routing decisions based on cost, using an AGI) Call transfer is working fine, the handsets have a Flash/R key to accomplish this. Using the Flash lets you start a second call, and once answered you can easily conference the second party in (softkey right on the screen), or transfer the call to the other party (via menu, then transfer). Using this capability, someone on a call can easily confer with another party, and bridge them into the call. AFAIK it is not possible for someone to join an existing call easily. You'd have to implement that in asterisk's dialplan, not on the Gigaset phone. My understanding is that the C610IP has a few more features than the 510. I might've also read somewhere that the 510 is obsolete. Can't find that link right now, but search mgraves.org (use the Gigaset tag to get some initial results). On Tue, Dec 11, 2012 at 2:37 PM, Roy Abshire r...@coopvr.com wrote: That is true about the A580. I don't like the interface much to check messages. Besides that every time I go to dial a number...it always uses the first digit pressed to go into phone mode..so I have to press the first digit twice... I would test other phones but it's for home and I can't fork over $$ to try them all out I have tested some Nokia cell phones, the N97, N900, and E71 and the E71 and N900 worked well. I didn't like the N97. Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 12:52 PM, Pete Mundy wrote: One thing I dislike about the A580H is that the handset always says 'You have new messages' if I've missed a call. It wouldn't bug me if it said 'missed call' but it tells me I have new messages and even lights up a red LED under a button with a picture of an envelope on it. I'm about to test an A510IP and an A610IP to compare against the A580. Fingers crossed neither of them has that issue, because the Gigaset phone is a pretty good phone other than that, and the difficulty doing a (blind) transfer, as referred to by the OP. Pete On 12/12/2012, at 8:57 AM, Roy Abshire r...@coopvr.com wrote: I've been using the Gigaset A580 Base and A58H Phone for about 3 years now. Never gave me problems. The call Quality is excellent! I only have 1 handset connected to the Base but I want more. I bought a Linksys WIP330 as a 2nd phone to try out and that works just as good without a base unit. The A580 Base supports up to 6 handsets. I have 6 Incoming VOIP Numbers using separate SIP accounts pointed to 1 Handset but you can point each SIP to separate handsets. The call goes to the first phone that picks up. When on a call, picking up another phone makes a separate call and does not conference. I don't use conference yet but I know you have to put the call on hold or something. The thing I don't like about the A580 and might be the same on all of them is that you can only specify 1 Sip
Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715
Pete, Thanks for testing these out. You said you like the 510 vs 580. Which one is newer? I have the 580. I'm staying tuned for your review of the 610. The 580 isn't DECT compatible and only supports some of the Gigaset handsets. I might sell mine and upgrade depending on what you learn about it. Can you tell me a little more about what you are testing for? Quality? Simultaneous In/Outgoing Calls using 1 or Multiple SIP accounts. Conference Calling? Any other Features? On 12/11/2012 08:17 PM, Pete Mundy wrote: At the risk of continuing off-topic conversation... Oh the M9 has it's own issues, don't you worry (not to mention it's _way_ more expensive than the Gigaset range)! I've been testing the A510 today and I've decided I like it more than the A580. The software (via the web interface) looks more polished and overall thus far it has performed fairly well (with no annoying 'new message' statement!). I haven't tried multiple SIP registrations with dual handsets yet, but I intend to. You guys realise these are standardised DECT handsets right? Ie that's where the 6-handset limitation comes from, and you should theoretically be able to register other manufacturers' handsets too. I'm about to try the C610IP model that Kai-Uwe mentioned (over the next few days). I think I'll like even more than the A580 since it has a higher resolution colour display. And even then it's still miles cheaper than the M9! Don't get me wrong, I do really like Snom phones too, although the M9 hasn't been entirely smooth sailing. I love the 320 desk phone (at the cheaper end of the scale, it has a great feel to it). Isn't it great that they all interact with Asterisk so well (there you go, I'm on-topic again ;-). Pete On 12/12/2012, at 4:12 PM, Mitul Limbani mi...@enterux.in wrote: Mebbe you guys should try snom m9 dect ip phone, i have been using it since over 3 years now without any of these issues. Mitul On Dec 12, 2012 4:25 AM, Kai-Uwe Jensen kujen...@gmail.com wrote: Using a Gigaset C610IP here, and am very happy with the features. The base station can handle two concurrent SIP calls, and another internal one at that. It does it with a single SIP registration to each server. You can setup multiple servers if you want to and define dial patterns/plans that determine which server gets used. After some playing around with it, I'm now using my setup connected to a single asterisk only. (Let asterisk make call routing decisions based on cost, using an AGI) Call transfer is working fine, the handsets have a Flash/R key to accomplish this. Using the Flash lets you start a second call, and once answered you can easily conference the second party in (softkey right on the screen), or transfer the call to the other party (via menu, then transfer). Using this capability, someone on a call can easily confer with another party, and bridge them into the call. AFAIK it is not possible for someone to join an existing call easily. You'd have to implement that in asterisk's dialplan, not on the Gigaset phone. My understanding is that the C610IP has a few more features than the 510. I might've also read somewhere that the 510 is obsolete. Can't find that link right now, but search mgraves.org (use the Gigaset tag to get some initial results). On Tue, Dec 11, 2012 at 2:37 PM, Roy Abshire r...@coopvr.com wrote: That is true about the A580. I don't like the interface much to check messages. Besides that every time I go to dial a number...it always uses the first digit pressed to go into phone mode..so I have to press the first digit twice... I would test other phones but it's for home and I can't fork over $$ to try them all out I have tested some Nokia cell phones, the N97, N900, and E71 and the E71 and N900 worked well. I didn't like the N97. Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 12:52 PM, Pete Mundy wrote: One thing I dislike about the A580H is that the handset always says 'You have new messages' if I've missed a call. It wouldn't bug me if it said 'missed call' but it tells me I have new messages and even lights up a red LED under a button with a picture of an envelope on it. I'm about to test an A510IP and an A610IP to compare against the A580. Fingers crossed neither of them has that issue, because the Gigaset phone is a pretty good phone other than that, and the difficulty doing a (blind) transfer, as referred to by the OP. Pete On 12/12/2012, at 8:57 AM, Roy Abshire r...@coopvr.com wrote: I've been using the Gigaset A580 Base and A58H Phone for about 3 years now. Never gave me problems. The call Quality is excellent! I only have 1 handset connected to the Base but I want more. I bought a Linksys WIP330 as a 2nd phone to try out and that works just as good without a base unit. The A580 Base supports up to 6 handsets. I have 6 Incoming VOIP Numbers
