[asterisk-users] No Codecs and app
Hi list, recently install asterisk 1.4.21 in a centos 5, and after having installer the zaptel 1.4.10.1 and libpri 1.4.4 I don't see in the directory module any codec, and neither app. almost install all the asterisk options this worries to me ! alone I see these packages inside the directory app_addon_sql_mysql.so cdr_addon_mysql.so res_config_mysql.so app_saycountpl.so chan_ooh323.so format_mp3.so some help that they can provide me? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Centile ipbx, anyone heard of this?
On Tue, Jun 24, 2008 at 5:58 AM, C. Savinovich [EMAIL PROTECTED] wrote: To be fair, Centile is better geared than asterisk for virtual pbx hosting. It comes with a system to manage virtual pbxs... it also handles the provisioning of most ip phones adequately, it is a totally different pbx although linux based. Interesting. Yes, it has a few phones it knows how to provision. I am using generic SIP device for both the phones currently in use. Although I don't know the details of your setup, it would not surprise me to see Centile accepting 2 different phones with the same extension on the same pbx. Well, my 4AM brainstorm didn't help. The phone I'm having trouble with is my favorite one, a Siemens S675IP. It is registered and works perfectly with 5 other SIP providers. On the Centile pbx, it can make calls but it can not be called. The web admin interface shows the correct public and NAT ip addresses and shows the phone in service. Calling it from another phone rings once and then goes to congestion, or at least that's the signal I hear. (It's wierd not being able to ssh in and see what's happening.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play sound on a specific channel
any idea? On Sat, Jun 14, 2008 at 9:50 AM, nik600 [EMAIL PROTECTED] wrote: Hi to all can i play a sound or a dtmf tone on a specific channel using AMI? Thanks to all -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] retrieve the status of a sip user using AMI
Hi to all. How can i retrieve the status of a user using the subscription? For example, if i use: exten = 200,hint,SIP/200 exten = 200,1,Dial(SIP/200) After that, how can i retrieve the status of the SIP/200 user using AMI ? Thanks to all in advance -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Kirk 600v3 Server with sip secret
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I'm trying to connect a kirk 4040 and a 4020 via a kirk server 600v3 to my asterisk (1.4.20) On the 600v3 I use latest radio and ip firmware. My question is, is it possible to set a secret in sip.conf for these two phones? Cause I had no success until now. If I comment the secret=x in sip.conf for that user, the phone registers. If I set a secret the server sends a 401 back, but the 600v3 doesn't resent the Register packet with the secret in it. Anyone discovered that too? This is my config for the phone: sip.conf [4333] nat=yes secret=4333 login=4333 callerid=Christoph Fuerstaller4333 call-limit=10 setvar=intern=1 callgroup=2 pickupgroup=2 [EMAIL PROTECTED] language=de disallow=all allow=g729 allow=gsm allow=ulaw allow=alaw type=friend host=dynamic dtmfmode=RFC2833 canreinvite=no qualify=yes context=intern subscribecontext=blf_group DECT user: Long Name: 4333 Name: 4333 Number: 4333 Password: 4333 Display Text: Christoph Fuerstaller I hope, anyone can help me with that. Chris... - -- COMMPANY | dialog solutions Franz-Josef-Strasse 33/4/43, 5020 Salzburg Tel: +43 662 879512 Fax: +43 662 875960 IP-Tel: +43 780 commpany (26667269) Email: [EMAIL PROTECTED] sip: [EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.9 (GNU/Linux) iEYEARECAAYFAkhgmEwACgkQR0exH8dhr/ajPwCfdIp5ums9laW4vhHcjhCKnTNv H/UAoLBHtCaAxQXG/EyecJs9TuW4CPMz =v/jO -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Loose connection with MySql.
Hello, I configured asterisk to use mysql for CDR. Well when i check from time to time I realize that asterisk loose connection with mysql (i use phpmyadmin and i watch the processes). Can anybody tell me how can i solve that problem? I want to have all cdr statistics logged in mysql, is very important for billing. Thank you for support. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GXW4024
Hi guys,
I'm testing the new gxw-4024 appliance but have a problem with attended
transfer, it works but after that the phone transfered the call, it
results busy for 60 seconds.
In my scenario the phone connected to 4024 (phone B) receive a call from
another sip client logged on asterisk server (phone A), it put it on
hold by pressing R (flash button) and dial another sip client also
logged on my asterisk (phone C). This one speak with B and accepts the
call. At this point, B hangs up by putting down the handset and let A
speaks with C.
I registered the port1 on asterisk server configured as follow (sip.conf
and extensions.conf ):
[207]
type = friend
username = password
host = dynamic
nat = never
port = 5060
context = per_tutti
secret = 207
dtmfmode = inband
canreinvite = yes
language = it
canreinvite = yes
mailbox = 207
qualify = yes
callerid = Test 207
[local]
exten = _[24]XX,1,Macro(exten,${EXTEN})
exten = _[24]XX,2,HangUp
[macro-exten]
exten = s,1,Dial(${ARG1})
exten = s,2,GoTo(s-${DIALSTATUS},1)
exten = s-BUSY,1,Busy()
exten = s-BUSY,2,HangUp
exten = s-NOANSWER,1,Congestion()
exten = s-NOANSWER,2,HangUp
exten = s-CONGESTION,1,Congestion()
exten = s-CONGESTION,2,HangUp
exten = s-CANCEL,1,Congestion()
exten = s-CANCEL,2,HangUp
As anyone tried similar scenario?
Thanks all
Giordano Grandis
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Re: [asterisk-users] Loose connection with MySql.
On 09:54, Tue 24 Jun 08, Catalin S. wrote: Hello, I configured asterisk to use mysql for CDR. Well when i check from time to time I realize that asterisk loose connection with mysql (i use phpmyadmin and i watch the processes). Can anybody tell me how can i solve that problem? I want to have all cdr statistics logged in mysql, is very important for billing. Thank you for support. Use cdr_adaptive_odbc backport for 1.4. That one does a check if the connection is still working, and if not it will reconnect. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Softphone accepting sip messages
Hello all, Someone knows any softphone which accept messages using sipsak? I just tried X-Lite and portsip without success Thanks Voipcrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone accepting sip messages
On 24 Jun 2008, at 09:29, voip crazy wrote: Hello all, Someone knows any softphone which accept messages using sipsak? I just tried X-Lite and portsip without success Thanks Voipcrazy. Take a look at firefly http://www.freshtel.net/download/internetphone/ I'm pretty sure it does. Otherwise all the IAX softphones will display IAX text frames. (including ours - www.phonefromhere.com). If you use IAX, the frame will have to come via asterisk not some arbitrary 3rd party running sipsak, which is less convenient but much more secure. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue with different music for each caller
Hi, is there an possibilty to have for each caller different music when queued. I see there only the global musiconhold = default in queues.conf, what menas same musci for all waiting callers. Any other idea to realize this? best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loose connection with MySql.
errr -you mean Asterisk doesn't ALWAYS check this and reconnect with the database ?!? WTF Since the CDRs are the literal Cash and Life Blood of many application why the heck would it NOT do this as part of its minimal basic operation ??? If it Doesn't do this for CDRs does it NOT do it for RealTime ?? If not, one could it up,screwed,blued and tatoed Is this functionality or lack there of documented anyplace ??? Michiel van Baak wrote: On 09:54, Tue 24 Jun 08, Catalin S. wrote: Hello, I configured asterisk to use mysql for CDR. Well when i check from time to time I realize that asterisk loose connection with mysql (i use phpmyadmin and i watch the processes). Can anybody tell me how can i solve that problem? I want to have all cdr statistics logged in mysql, is very important for billing. Thank you for support. Use cdr_adaptive_odbc backport for 1.4. That one does a check if the connection is still working, and if not it will reconnect. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Command Option D Early Bridged
How do other applications, such a the automated dialers from telemarketers, reliably detect when the call has been answered ? I thought this sort of basic functionality that had been around for quite awhile. Jared Smith wrote: On Thu, 2008-06-12 at 16:43 +0800, tcchan wrote: However, in my experience, the timing the call get bridged is not consistance, Do you happen to be calling out over an analog phone line? In the case of dialing out an analog line, we have no easy way of knowing when the far-end has answered the call, so the call is considered answered at the time the call is dialed. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TRANSFER_CONTEXT ignored?
The TRANSFER_CONTEXT has only ever worked for blind transfers for me and gets ignored for attended transfers. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48)
Did I add this yet? 2008/3/22 Faraz Khan [EMAIL PROTECTED]: Just checked 1.6.0beta4 - the res_ldap.conf file still has PBX* attributes - which I'm guessing would be confusing to any new user. the schema file looks file though, the missing voicemail/queue part is what we have added. Quoting Faraz Khan [EMAIL PROTECTED]: Did you manage to upload those changes? Some of your schema/ldif files were deleted by the bug admin. You might want to upload them at voip-info Furthermore, the multi_ldap call is broken in res_config_ldap.c - I even started a bounty on it but looks like few people are interested and/or bounty amount is too low :) without the multi_ldap fix, all we can realistically do is put sip.conf in ldap- which is a decent improvement however it would be amazing if the entire dialplan/queues/etc could be put into voicemail as well. Right now one has to use LDAP for account and Mysql for extensions/queues. Quoting Gavin Henry [EMAIL PROTECTED]: On 17/03/2008, Faraz Khan [EMAIL PROTECTED] wrote: Good Idea and done. It is now available here: http://www.voip-info.org/wiki/view/LDAP The correct LDAP Schema is included: /asterisk-1.6.0-beta4/contrib/scripts/asterisk.ldap-schema and /asterisk-1.6.0-beta4/contrib/scripts/asterisk.ldif Good work though. I'm just uploading some fixes to it at: http://bugs.digium.com/view.php?id=12177 Gavin. -- http://www.suretecsystems.com/services/openldap/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://www.suretecsystems.com/services/openldap/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP
That's excellent! So in theory one could not make Asterisk compatible SIP softphone in Flash (since Flash only supports TCP). Nice... BR, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Tuesday, June 24, 2008 1:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP over TCP No, TCP for media as well. I though that was the whole point of SIP over TCP. Michael On Mon, 23 Jun 2008 16:59:00 +0200, Asterisk wrote: Hi, But you can only route SIP signalization over TCP. Audio stream must still go thru UDP, right? BR, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Sunday, June 22, 2008 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP over TCP On 6/22/08, Michael Graves [EMAIL PROTECTED] wrote: Ok, so now that it's possible to implement SIP over TCP instead of UDP why would I want to do this? Beyond simply integration with M$ OCS. And what are the implications for management of QoS? I would expect that lost packets would be less of a factor. Thanks, Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] Michael, The main advantages for SIP over TCP that I know of (in no particular order): - Better compatibility with NAT devices (it seems some of them don't do UDP well) - Support for TLS - Support for packet fragmentation (to support large/diverse SDPs, headers, etc) I'm sure there are other ones but that's all I can think of this early on a Sunday morning... -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIfTime Function
I googled some information on voip.org. Its my fault though and implemented the sample implementation without creating the context an the include statements. On Mon, Jun 23, 2008 at 10:33 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: If any docs were the cause of this (very important) misconception, maybe the docs could be reworded. Do you remember what caused you to think that context was created automatically? broadband Voice wrote: fc7234153*CLI dialplan show open There is no existence of 'open' context I was under the impression that this was part of the Asterisk default libraries. I will create the context then and also add the include files. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Lockups with IAX2 and cdr_odbc in 1.4.21 is it my weird config ?
Is anyone using 1.4.21 with cdr_odbc and IAX channels successfully? I'm getting lockups where asterisk stops responding (to anything). Foolishly I've built a box with 2 new things on it, 1.4.21 and Oracle as the odbc server. If others are running it fine against MySql or postgres , I'll focus on the oracle side. I was just wondering if it was a side effect of the new IAX threading in 1.4.21. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lockups with IAX2 and cdr_odbc in 1.4.21 is it my weird config ?
On 12:44, Tue 24 Jun 08, Tim Panton wrote: Is anyone using 1.4.21 with cdr_odbc and IAX channels successfully? I'm getting lockups where asterisk stops responding (to anything). Foolishly I've built a box with 2 new things on it, 1.4.21 and Oracle as the odbc server. If others are running it fine against MySql or postgres , I'll focus on the oracle side. I was just wondering if it was a side effect of the new IAX threading in 1.4.21. Looks like this one: Issue 12925 [Channels/chan_iax2] IAX2 channel gets stuck, causes CLI to get stuck, * won't restart http://bugs.digium.com/view.php?id=12925 -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Codecs and app
On June 24, 2008 01:57:45 am troxlinux wrote: Hi list, recently install asterisk 1.4.21 in a centos 5, and after having installer the zaptel 1.4.10.1 and libpri 1.4.4 I don't see in the directory module any codec, and neither app. almost install all the asterisk options this worries to me ! alone I see these packages inside the directory app_addon_sql_mysql.so cdr_addon_mysql.so res_config_mysql.so app_saycountpl.so chan_ooh323.so format_mp3.so some help that they can provide me? Looks like you have installed asterisk-addons and not asterisk itself... if you compiled from source maybe you just forgot to 'make install' for asterisk? -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loose connection with MySql.
On Tuesday 24 June 2008 01:54:11 Catalin S. wrote: Hello, I configured asterisk to use mysql for CDR. Well when i check from time to time I realize that asterisk loose connection with mysql (i use phpmyadmin and i watch the processes). Can anybody tell me how can i solve that problem? I want to have all cdr statistics logged in mysql, is very important for billing. Are you actually missing CDRs, or are you just concerned that you MIGHT be? The only reason that you might be missing CDRs from the MySQL backend is if the MySQL server was completely down (i.e. not just disconnected from Asterisk, but completely down) at the time that the CDR was posted. All SQL insertions (for this backend) are automatically retried if the initial query failed. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lockups with IAX2 and cdr_odbc in 1.4.21 is it my weird config ?
On 24 Jun 2008, at 13:09, Michiel van Baak wrote: On 12:44, Tue 24 Jun 08, Tim Panton wrote: Is anyone using 1.4.21 with cdr_odbc and IAX channels successfully? I'm getting lockups where asterisk stops responding (to anything). Foolishly I've built a box with 2 new things on it, 1.4.21 and Oracle as the odbc server. If others are running it fine against MySql or postgres , I'll focus on the oracle side. I was just wondering if it was a side effect of the new IAX threading in 1.4.21. Looks like this one: Issue 12925 [Channels/chan_iax2] IAX2 channel gets stuck, causes CLI to get stuck, * won't restart http://bugs.digium.com/view.php?id=12925 -- Thanks! There was I blaming oracle. I'll downgrade. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loose connection with MySql.
On Tuesday 24 June 2008 04:43:19 Al Baker wrote: errr -you mean Asterisk doesn't ALWAYS check this and reconnect with the database ?!? WTF Since the CDRs are the literal Cash and Life Blood of many application why the heck would it NOT do this as part of its minimal basic operation ??? If it Doesn't do this for CDRs does it NOT do it for RealTime ?? If not, one could it up,screwed,blued and tatoed Is this functionality or lack there of documented anyplace ??? You might want to check your facts before launching into a diatribe. Both the MySQL backend driver for CDR as well as the MySQL backend driver for Realtime reconnect if possible during a query. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lockups with IAX2 and cdr_odbc in 1.4.21 is it my weird config ?
On Tuesday 24 June 2008 06:44:21 Tim Panton wrote: Is anyone using 1.4.21 with cdr_odbc and IAX channels successfully? I'm getting lockups where asterisk stops responding (to anything). Foolishly I've built a box with 2 new things on it, 1.4.21 and Oracle as the odbc server. If others are running it fine against MySql or postgres , I'll focus on the oracle side. I was just wondering if it was a side effect of the new IAX threading in 1.4.21. I'm not aware of anything that would cause conflicts between the threading in IAX and the locking in cdr_odbc. However, if you're getting a lockup, try recompiling with DONT_OPTIMIZE and DEBUG_THREADS, then get the output of 'core show locks' when the problem occurs and open an issue on http://bugs.digium.com with that output uploaded as a file. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Command Option D Early Bridged
On Tue, 2008-06-24 at 05:54 -0400, Al Baker wrote: How do other applications, such a the automated dialers from telemarketers, reliably detect when the call has been answered ? I thought this sort of basic functionality that had been around for quite awhile. For digital connections (such as a PRI on a T1 or E1), the telco sends a signal when the far end has answered the call. For analog lines (at least here in the United States), however, there is typically no signaling from the telco to let you know when the far end has answered the call. The best you can do is to try to listen to the audio and use audio characteristics to try to determine if it was answered. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lockups with IAX2 and cdr_odbc in 1.4.21 is it my weird config ?
On 24 Jun 2008, at 13:27, Tilghman Lesher wrote: On Tuesday 24 June 2008 06:44:21 Tim Panton wrote: Is anyone using 1.4.21 with cdr_odbc and IAX channels successfully? I'm getting lockups where asterisk stops responding (to anything). Foolishly I've built a box with 2 new things on it, 1.4.21 and Oracle as the odbc server. If others are running it fine against MySql or postgres , I'll focus on the oracle side. I was just wondering if it was a side effect of the new IAX threading in 1.4.21. I'm not aware of anything that would cause conflicts between the threading in IAX and the locking in cdr_odbc. However, if you're getting a lockup, try recompiling with DONT_OPTIMIZE and DEBUG_THREADS, then get the output of 'core show locks' when the problem occurs and open an issue on http://bugs.digium.com with that output uploaded as a file. A bit more experimentation and I've learnt the following: 1) it isn't present in 1.4.4 2) it still happens in 1.4.21 if I disable cdr_odbc - but less often. 3) it freezes the CLI - so I don't think 'core show locks' will work. 4) it (or something similar) is in http://bugs.digium.com/view.php?id=12925 The machine isn't in production (yet) so if there are other tests I can run I'd be happy to. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP
On Tue, 24 Jun 2008 12:36:52 +0200, Asterisk wrote: That's excellent! So in theory one could not make Asterisk compatible SIP softphone in Flash (since Flash only supports TCP). Nice... BR, Alex I beleive that this hs already been done, although I can't recall by whom. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Tuesday, June 24, 2008 1:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP over TCP No, TCP for media as well. I though that was the whole point of SIP over TCP. Michael On Mon, 23 Jun 2008 16:59:00 +0200, Asterisk wrote: Hi, But you can only route SIP signalization over TCP. Audio stream must still go thru UDP, right? BR, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Sunday, June 22, 2008 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP over TCP On 6/22/08, Michael Graves [EMAIL PROTECTED] wrote: Ok, so now that it's possible to implement SIP over TCP instead of UDP why would I want to do this? Beyond simply integration with M$ OCS. And what are the implications for management of QoS? I would expect that lost packets would be less of a factor. Thanks, Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] Michael, The main advantages for SIP over TCP that I know of (in no particular order): - Better compatibility with NAT devices (it seems some of them don't do UDP well) - Support for TLS - Support for packet fragmentation (to support large/diverse SDPs, headers, etc) I'm sure there are other ones but that's all I can think of this early on a Sunday morning... -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue with different music for each caller
Hello Thomas
you can use different music for each caller if you like.
in extensions.conf you can set the music class.
exten = s,n,Set(CHANNEL(musicclass)=yourmusicforthiscaller)
and if you like different music for each caller you can set a variable with
the musicname and then set the musicclass:
exten = s,n,Set(mymusicclass=${CALLERID(num)}musicclass)
exten = s,n,Set(CHANNEL(musicclass)=${mymusicclass})
so you set the music for each calling number
when the caller has the number 1234 the music will be 1234musicclass
when the caller has the cid 456 the music will be 456musicclass
you see, everything is possible.
hope to help,
Martin
- Original Message -
From: Thomas Winter [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, June 24, 2008 11:22 AM
Subject: [asterisk-users] Queue with different music for each caller
Hi,
is there an possibilty to have for each caller different music when queued.
I see there only the global musiconhold = default in queues.conf, what menas
same musci for all waiting callers.
Any other idea to realize this?
best regards
Thomas
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Re: [asterisk-users] Centile ipbx, anyone heard of this?
Of course you can ssh. And you can trace whats going on at 3 different levels. You can also open a trouble ticket with them. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Tuesday, June 24, 2008 2:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Centile ipbx, anyone heard of this? On Tue, Jun 24, 2008 at 5:58 AM, C. Savinovich [EMAIL PROTECTED] wrote: To be fair, Centile is better geared than asterisk for virtual pbx hosting. It comes with a system to manage virtual pbxs... it also handles the provisioning of most ip phones adequately, it is a totally different pbx although linux based. Interesting. Yes, it has a few phones it knows how to provision. I am using generic SIP device for both the phones currently in use. Although I don't know the details of your setup, it would not surprise me to see Centile accepting 2 different phones with the same extension on the same pbx. Well, my 4AM brainstorm didn't help. The phone I'm having trouble with is my favorite one, a Siemens S675IP. It is registered and works perfectly with 5 other SIP providers. On the Centile pbx, it can make calls but it can not be called. The web admin interface shows the correct public and NAT ip addresses and shows the phone in service. Calling it from another phone rings once and then goes to congestion, or at least that's the signal I hear. (It's wierd not being able to ssh in and see what's happening.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chef-secretary scenario
Hi all, I'm trying to implement such a scenario where the Chef picks up his phone and his secretary can see that he is busy. Something like blf, I guess. But so far I've only managed to notify the secretary that the chef is receiving a call. I want to do it the other way around though. I'd like for her to see in her phone, the light corresponding to the chef's extension light up whenever he uses the phone (also when he picks it up if that's possible). So she should always know when he's busy. Is there a way to do that? Thanks, David ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lockups with IAX2 and cdr_odbc in 1.4.21 is it my weird config ?
On Tuesday 24 June 2008 07:46:33 Tim Panton wrote: On 24 Jun 2008, at 13:27, Tilghman Lesher wrote: On Tuesday 24 June 2008 06:44:21 Tim Panton wrote: Is anyone using 1.4.21 with cdr_odbc and IAX channels successfully? I'm getting lockups where asterisk stops responding (to anything). Foolishly I've built a box with 2 new things on it, 1.4.21 and Oracle as the odbc server. If others are running it fine against MySql or postgres , I'll focus on the oracle side. I was just wondering if it was a side effect of the new IAX threading in 1.4.21. I'm not aware of anything that would cause conflicts between the threading in IAX and the locking in cdr_odbc. However, if you're getting a lockup, try recompiling with DONT_OPTIMIZE and DEBUG_THREADS, then get the output of 'core show locks' when the problem occurs and open an issue on http://bugs.digium.com with that output uploaded as a file. A bit more experimentation and I've learnt the following: 1) it isn't present in 1.4.4 2) it still happens in 1.4.21 if I disable cdr_odbc - but less often. 3) it freezes the CLI - so I don't think 'core show locks' will work. There are very few things that can actually freeze the CLI. I suspect what is more likely is that the last command executed on the CLI depends upon a lock that it cannot get, and since the CLI runs commands synchronously, it would appear that everything is locked up. You should still be able to get a 'core show locks' by doing: asterisk -rx 'core show locks' as that will ensure that no other commands are run before this on that particular remote console. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Openfire Asterisk-IM Plugin Performance Observation
Is it better for production to run Openfire on a separate server than the PBX? Since discovering linux vservers I put every service into its own install. Each install can be very lightweight and vservers only add about 1MB to ram usage (I don't run a separate init process), so very lightweight. The advantage is that it's super simple to backup each server and you can test upgrades by simply copying the image, fire up a new instance, test your upgrade, then burn it down again... Piece of cake to shuffle services between real machines also (preserving IP addresses also if that's required). Backups can be done very easily (make the /vserver dir an LVM disk) Good luck Ed W ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GSM Gateway Project
Hi There http://www.2n.cz/company/2n_history.html offer this kind of products. they works vey well with asterisk. Ciao Andrea Michael Graves ha scritto: On Mon, 23 Jun 2008 10:09:21 -0400, Steve Totaro wrote: On Mon, Jun 23, 2008 at 9:57 AM, [EMAIL PROTECTED] wrote: The quad-band model is around $250 USD. See Ebay auction here http://tinyurl.com/5tvoa9 Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] skype mjgraves FWD 54245 Do any of these do SMS? Yes, most of them do. See #7 on the following page: http://www.portech.com.tw/eweb/index1.htm Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with Nextone using H323
Hi people, Someone have already used asterisk with Nextone? I`m trying to use it, but there are some problems.. One of these are when we set up a connection between Nextone and Asterisk using H323, we use our asterisk server as a Softswitch in the Nextone configuration, so it doesn`t work. But, when we just change (in Nextone configuration) from Softswitch to Gateway, it work. Where is the difference? I`m using chan_ooh323 in my asterisk server. This is my ooh323.conf: - [general] ;Default - 1720 ;port=1720 bindaddr= IP_ADDRESS ;This parameter indicates whether channel driver should register with ;gatekeeper as a gateway or an endpoint. ;Default - no ;gateway=yes ;Whether asterisk should use fast-start and tunneling for H323 connections. ;Default - yes ;faststart=no ;h245tunneling=no ;H323-ID to be used for asterisk server ;Default - Asterisk PBX h323id=GW2 e164=1521# ;CallerID to use for calls ;Default - Same as h323id callerid=MediaXChange 1.0 ;Whether this asterisk server will use gatekeeper. ;Default - DISABLE ;gatekeeper = DISCOVER ;gatekeeper = a.b.c.d ;gatekeeper = 189.44.163.125 logfile=/var/log/asterisk/h323_log context=default disallow=all allow=g729 allow=g723 dtmfmode=rfc2833 - Anyone know what can I do? or what am I doing wrong??? Thank`s a lot for the opportunity. Everton Goularth ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Centile ipbx, anyone heard of this?
On Tue, Jun 24, 2008 at 3:22 PM, C. Savinovich [EMAIL PROTECTED] wrote: Of course you can ssh. And you can trace whats going on at 3 different levels. You can also open a trouble ticket with them. No, I don't have the auth to ssh, but in the end it was a config error on my end. The fact that they patiently worked through this (there was a ticket, I just like to get more input from experienced people) and that they got it running confirms I made the right choice of people to work with. Great service and mea culpa, I am not worthy. I really wondered why a phone would work with every other provider and asterisk 1.2 and not on this thing. thx again for the input. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Nextone using H323
El mar, 24-06-2008 a las 12:20 -0300, Everton Goularth escribió: I`m using chan_ooh323 in my asterisk server. This is my ooh323.conf: Have you tried with chan_h323.so? I've one gateways that uses h.323 and works only with chan_h323.so . Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk!
This is just a note that the fixes in the CDRfix4 and CDRfix6 branches are getting closer to being merged into 1.4, trunk, and 1.6.x. If CDR's are important to you, and you ignore this notice, then you deserve what you get! These branches address various long-standing bugs, most of which are regressions from 1.2. It is hoped that these fixes will solve most of the problems introduced by the various changes made in 1.4 and trunk, without losing the fixes made in those changes. To test out these branches, you can: svn co http://svn.digium.com/svn/asterisk/team/murf/CDRfix4 or svn co http://svn.digium.com/svn/asterisk/team/murf/CDRfix6 The above commands will create a directory called CDRfix4 (or CDRfix6) in the current directory, which contains an entire copy of the asterisk source. You can cd into this dir and do the configure/make menuselect/ make/make install thing there to your hearts content. The CDRfix4 branch is based on 1.4; The CDRfix6 branch is based on trunk (which is still close enough to 1.6.0 that it won't take much effort to merge it 1.6.0 also.) The bugs that will hopefully be addressed are: http://bugs.digium.com/view.php?id=10927 http://bugs.digium.com/view.php?id=11093 http://bugs.digium.com/view.php?id=12724 http://bugs.digium.com/view.php?id=12907 and perhaps others. The goal was to restore the code roughly to 1.2 behavior when it came to transfers, minus any bad behavior that 1.2 had. So, entire legs missing from transfers, missing or bad times, etc, seem to mostly solved. The fixes do NOT fulfil requests to further subdivide CDR's in xfer situations, as I'm not warm and fuzzy on a general consensus as to exactly what the new CDR's would say. I haven't been able to engage really anyone in getting details ironed out on these issues. Folks have made suggestions, good ones at that, but everyone seems to be of a mind that before we extend or upgrade the current CDR system, we should produce a specification, and see if the community can come to a consensus on that spec. So, I think I might make a proposal for enhancement of the existing CDR's to give more details about xfer situations, and we can hash out the details from there. This proposal can then serve as the spec for future enhancements. Also, keep in mind, that we have a new facility being groomed for merging into trunk: http://svn.digium.com/svn/asterisk/team/group/newcdr which will introduce some new concepts that will help in forming billing records; it is single-event based, and introduces a new channel value, linkedid, which is spread between channels that 'interact', thus allowing you to more easily collect events that are related via transfers, conferences, parking, holding, etc. So, please, please, please, test these branches against your implementations, and report any problems you see, so we can solve problems before they get merged! Problems and complaints can be added to the bugs mentioned above, choose the one that seems most closely related to the problem you are having. murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calls drop + Didn't get a frame from channel log message
Hi, sometimes Asterisk drops calls and shows Didn't get a frame from channel in its log file. Unfortunately Google gives no answers even if a lot of people ask for help. A fast look into the code shows Asterisk entering a loop where voice is been transferred and every loop Asterisk waits for a frame, exiting the loop if no frame has arrived. It seems to be a problem not depending on the kind of channel...happens with ISDN and PRI lines. What is stopping the frames, making Asterisk exiting that loop and dropping the calls? Thank you. Giorgio. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk!
We should merge this changes immediately. At least, fix NoCDR(), which really affects the business. Maybe you can do that meanwhile. I know your changes work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Murphy Sent: Tuesday, June 24, 2008 11:28 AM To: asterisk-users Cc: Asterisk Developers Mailing List Subject: [asterisk-dev] Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk! This is just a note that the fixes in the CDRfix4 and CDRfix6 branches are getting closer to being merged into 1.4, trunk, and 1.6.x. If CDR's are important to you, and you ignore this notice, then you deserve what you get! These branches address various long-standing bugs, most of which are regressions from 1.2. It is hoped that these fixes will solve most of the problems introduced by the various changes made in 1.4 and trunk, without losing the fixes made in those changes. To test out these branches, you can: svn co http://svn.digium.com/svn/asterisk/team/murf/CDRfix4 or svn co http://svn.digium.com/svn/asterisk/team/murf/CDRfix6 The above commands will create a directory called CDRfix4 (or CDRfix6) in the current directory, which contains an entire copy of the asterisk source. You can cd into this dir and do the configure/make menuselect/ make/make install thing there to your hearts content. The CDRfix4 branch is based on 1.4; The CDRfix6 branch is based on trunk (which is still close enough to 1.6.0 that it won't take much effort to merge it 1.6.0 also.) The bugs that will hopefully be addressed are: http://bugs.digium.com/view.php?id=10927 http://bugs.digium.com/view.php?id=11093 http://bugs.digium.com/view.php?id=12724 http://bugs.digium.com/view.php?id=12907 and perhaps others. The goal was to restore the code roughly to 1.2 behavior when it came to transfers, minus any bad behavior that 1.2 had. So, entire legs missing from transfers, missing or bad times, etc, seem to mostly solved. The fixes do NOT fulfil requests to further subdivide CDR's in xfer situations, as I'm not warm and fuzzy on a general consensus as to exactly what the new CDR's would say. I haven't been able to engage really anyone in getting details ironed out on these issues. Folks have made suggestions, good ones at that, but everyone seems to be of a mind that before we extend or upgrade the current CDR system, we should produce a specification, and see if the community can come to a consensus on that spec. So, I think I might make a proposal for enhancement of the existing CDR's to give more details about xfer situations, and we can hash out the details from there. This proposal can then serve as the spec for future enhancements. Also, keep in mind, that we have a new facility being groomed for merging into trunk: http://svn.digium.com/svn/asterisk/team/group/newcdr which will introduce some new concepts that will help in forming billing records; it is single-event based, and introduces a new channel value, linkedid, which is spread between channels that 'interact', thus allowing you to more easily collect events that are related via transfers, conferences, parking, holding, etc. So, please, please, please, test these branches against your implementations, and report any problems you see, so we can solve problems before they get merged! Problems and complaints can be added to the bugs mentioned above, choose the one that seems most closely related to the problem you are having. murf -- Steve Murphy Software Developer Digium ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue with different music for each caller
On Tuesday 24 June 2008 15:22, Martin Schrott - thinking:systems wrote: Hello Thomas you can use different music for each caller if you like. in extensions.conf you can set the music class. exten = s,n,Set(CHANNEL(musicclass)=yourmusicforthiscaller) Hi Martin, thanks for your suggestion, I forgot to notice that Iam still using 1.2.X Jun 24 17:45:31 ERROR[17784]: pbx.c:1437 ast_func_write: Function CHANNEL not registered So, this didnt work for me. best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chef-secretary scenario
To see? how? what phone do you use? Snoms imprement that, you got BLINKING and ON state BLINKING=calling or being called ON=on the phone 2008/6/24 Vazquez David [EMAIL PROTECTED]: Hi all, I'm trying to implement such a scenario where the Chef picks up his phone and his secretary can see that he is busy. Something like blf, I guess. But so far I've only managed to notify the secretary that the chef is receiving a call. I want to do it the other way around though. I'd like for her to see in her phone, the light corresponding to the chef's extension light up whenever he uses the phone (also when he picks it up if that's possible). So she should always know when he's busy. Is there a way to do that? Thanks, David ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chef-secretary scenario
On Tue, 24 Jun 2008, Vazquez David wrote: Hi all, I'm trying to implement such a scenario where the Chef picks up his phone and his secretary can see that he is busy. Something like blf, I guess. It's like BLF because that's exactly what it's for.. But so far I've only managed to notify the secretary that the chef is receiving a call. I want to do it the other way around though. I'd like for her to see in her phone, the light corresponding to the chef's extension light up whenever he uses the phone (also when he picks it up if that's possible). So she should always know when he's busy. Is there a way to do that? Yes. Configure hints in asterisk and BLF in the secretarys phone. I don't understand how you get notifications one way, but not the other though (unless you're doing it by not using BLF) What phones have you got? What do your hints look like in the extensions.conf file? You can't get the status of lifting the handset on a SIP phone though (well, not that I'm aware of) However, if it's an analogue phone on a TDM400 card (or equivalent, I guess) then it does work and the BLF LED on my Grandstream phone turns Red as soon as I take the analogue phone off-hook... So BLF is what you want, and optionally an analogue phone +TDM card for the boss... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] does asterisk 1.4.20 run on a 486 sx
I have compiled asterisk 1.4.20 on a 486 (sx) machine. No floating point but math emulation is used in the kernel. When I run asterisk -vc all I get is Illegal instruction. I compiled as normally I do. Whats my next step. this is download source and compiled. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] does asterisk 1.4.20 run on a 486 sx
On Tuesday 24 June 2008 12:39:36 Jerry Geis wrote: I have compiled asterisk 1.4.20 on a 486 (sx) machine. No floating point but math emulation is used in the kernel. When I run asterisk -vc all I get is Illegal instruction. I compiled as normally I do. Whats my next step. this is download source and compiled. Upgrade to SVN revision 123869 or higher. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] does asterisk 1.4.20 run on a 486 sx
Trash your PC is you next step :) 2008/6/24 Tilghman Lesher [EMAIL PROTECTED]: On Tuesday 24 June 2008 12:39:36 Jerry Geis wrote: I have compiled asterisk 1.4.20 on a 486 (sx) machine. No floating point but math emulation is used in the kernel. When I run asterisk -vc all I get is Illegal instruction. I compiled as normally I do. Whats my next step. this is download source and compiled. Upgrade to SVN revision 123869 or higher. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Centile ipbx, anyone heard of this?
On Tue, 24 Jun 2008 08:28:35 +0200, randulo wrote: On Tue, Jun 24, 2008 at 5:58 AM, C. Savinovich [EMAIL PROTECTED] wrote: To be fair, Centile is better geared than asterisk for virtual pbx hosting. It comes with a system to manage virtual pbxs... it also handles the provisioning of most ip phones adequately, it is a totally different pbx although linux based. Interesting. Yes, it has a few phones it knows how to provision. I am using generic SIP device for both the phones currently in use. Although I don't know the details of your setup, it would not surprise me to see Centile accepting 2 different phones with the same extension on the same pbx. Well, my 4AM brainstorm didn't help. The phone I'm having trouble with is my favorite one, a Siemens S675IP. It is registered and works perfectly with 5 other SIP providers. On the Centile pbx, it can make calls but it can not be called. The web admin interface shows the correct public and NAT ip addresses and shows the phone in service. Calling it from another phone rings once and then goes to congestion, or at least that's the signal I hear. (It's wierd not being able to ssh in and see what's happening.) Randy, This is exactly what was happening when I used an Aastra 480i CT with OnSIP. According to OnSIP it's not a supported phone, although the newer 57i CT does work with OnSIP. It seemed that the phone was losing registration with the provider. I was not able to overcome this in the phone or provider settings. My ultimate solution was to build a small Asterisk instance (Astlinux) on a thin client (HP T5700) and use it strickly as a bridge device for the phone. For whatever reason, the Astlinux box could sustain the registration and pass the incomming calls to the phone. This is very similar to another idea that I once had but never actually implemented. That is, using a small embedded Asterisk device as a SIPIAX2 protocol translator to facilitate complex NAT traversal. I thought that Astlinux on Gumstix hardware would be ideal for such a task. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Nextone using H323
Any reason in particular why you don't use SIP between your Asterisk and NexTone? This is how I have ours connected and it works well. The only issue I've experienced is that some of the carriers that only support g729 AB have trouble with the dtmf tones from g729A, but this is not SIP specific. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M. Sent: Tuesday, June 24, 2008 11:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk with Nextone using H323 El mar, 24-06-2008 a las 12:20 -0300, Everton Goularth escribió: I`m using chan_ooh323 in my asterisk server. This is my ooh323.conf: Have you tried with chan_h323.so? I've one gateways that uses h.323 and works only with chan_h323.so . Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Openfire Asterisk-IM Plugin Performance Observation
i used it on one server a little while ago. my primary use was ability to show each user's status on spark. i did not get consistence results, phone status was not accurate. and did not try it after that, maybe its fixed in newer versions. On Fri, Jun 20, 2008 at 2:44 PM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: See below: Erik Anderson wrote: On Fri, Jun 20, 2008 at 12:47 PM, JR Richardson [EMAIL PROTECTED] wrote: So now the PBX is over 1.2 Gig for the installation. Typical PBX installs are under 600 Meg. This makes me wonder about server stability, reliability and performance as uptime creeps on and user count increases over 50 to 100+. Increased data on the hard drive won't really have an affect on reliability or performance. Can anyone give me feedback on real world experience with this type of setup and any performance issues that my arise? I can't speak directly to the asterisk + openfire situation. I can, however, say that I've been running openfire for nearly a year now on a very highly-loaded server (other than openfire, it's running nagios and cacti, monitoring about 300 devices around our network) - the load average on this 5-year single processor old dell server is pegged near 1.00 24x7. I haven't had a single problem with openfire, and I have between 50 and 100 open sessions at any one time. In the year that I've been running openfire, I've only had to restart it once, and that was to upgrade the software. It takes very little CPU, and a modest amount of RAM. Is it better for production to run Openfire on a separate server than the PBX? What's your definition of better. Is it better to not have all your eggs in one basket? Is it better to only need to purchase one server? Is it better to only have one server to manage/update/etc versus two? My biggest concern is deploying a 100+ user environment with high call volume and high chat volume. Java seems to be a bit resource hungry with the user notifications and call pop ups. I would hate to have the IM server walking over Asterisk and affecting call quality or PBX stability. Speaking personally, I'd have no problems putting openfire and asterisk on the same box. If needed, you could even just nice the We run with the openfire process on the same box as the * server - we have not had a single problem with openfire in over 2 years now. openfire process down to a lower priority than asterisk - it's not as latency-sensitive as asterisk is. I'd doubt you'll need to do that, though. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GSM Gateway Project
I already checked them out. If you read their fine prints well they have minutes limitations then you have to buy licenses. From the responses that I got, I can get one the pci gsm cards with the drivers and that will work for us except that it does not scale very well. On Tue, Jun 24, 2008 at 10:58 AM, Andrea Cristofanini [EMAIL PROTECTED] wrote: Hi There http://www.2n.cz/company/2n_history.html offer this kind of products. they works vey well with asterisk. Ciao Andrea Michael Graves ha scritto: On Mon, 23 Jun 2008 10:09:21 -0400, Steve Totaro wrote: On Mon, Jun 23, 2008 at 9:57 AM, [EMAIL PROTECTED] wrote: The quad-band model is around $250 USD. See Ebay auction here http://tinyurl.com/5tvoa9 Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED][EMAIL PROTECTED] skype mjgraves FWD 54245 Do any of these do SMS? Yes, most of them do. See #7 on the following page: http://www.portech.com.tw/eweb/index1.htm Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] [EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mpg123 problem
On Sun, Jun 22, 2008 at 12:24:22AM -0700, fateme fatah wrote: I want to install mpg123-0.59r on my asterisk server.I downloaded it in /usr/src then untared it and I typed these command : Just have a look at www.mpg123.org and fetch the up to date version. 0.59r is probably available with your distribution but it is known to cause some problems. -- Stefan Tichy ( asterisk2 at pi4tel dot de ) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21 + Realtime Queues = Agents Not Ringing?
I recently observed a similar behaviour under 1.4.21. The member was a SIP phone which had its calls forwarded to another SIP phone via its built-in configuration... (fyi: linksys spa922) For some reason, asterisk could not manage this scenario. I still have to test it better to understand if this is supposed to work or not. Could that be your case ? (not very probable, I know...) -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loose connection with MySql.
Hello guys, thank you for all your answers. I'll will check and i keep you informed of what's happening next. Note that mysql and asterisk is on the same machine so is not a problem of connectivity or mysql machine to be down. On Tue, Jun 24, 2008 at 3:22 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 24 June 2008 04:43:19 Al Baker wrote: errr -you mean Asterisk doesn't ALWAYS check this and reconnect with the database ?!? WTF Since the CDRs are the literal Cash and Life Blood of many application why the heck would it NOT do this as part of its minimal basic operation ??? If it Doesn't do this for CDRs does it NOT do it for RealTime ?? If not, one could it up,screwed,blued and tatoed Is this functionality or lack there of documented anyplace ??? You might want to check your facts before launching into a diatribe. Both the MySQL backend driver for CDR as well as the MySQL backend driver for Realtime reconnect if possible during a query. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls drop + Didn't get a frame from channel log message
I have googled a lot to find solution to the same exact problem described in your message but no real solution yet. here is my config 1 physical network 25 pc windows 25 phones IP330 IP550 SIP 2.1.2 no vlan CDP disabled some with dhcp some with fixed ip to see if there is a diff 3 switchs connected to each others 1 cisco switch 35xx for pcs 2 linksys 24P P OE for phones 1 patton PRI gateway to isdn 1 asterisk server 1.12.18 talking sip to Patton , for each phone, type friend can re-invite no, nat no symptoms: call drop randomly , can be after 10 s or 2000 seconds ! same log didn't get frame etc fews drops per phones per day but very irritating for the customer :-( tried to power phones with adapters to avoid power pbs from the switch , same result if someone met this problem before get an idea to fix it , I wd appreciate ! thanks jl On Tue, Jun 24, 2008 at 5:37 PM, gincantalupo [EMAIL PROTECTED] wrote: Hi, sometimes Asterisk drops calls and shows Didn't get a frame from channel in its log file. Unfortunately Google gives no answers even if a lot of people ask for help. A fast look into the code shows Asterisk entering a loop where voice is been transferred and every loop Asterisk waits for a frame, exiting the loop if no frame has arrived. It seems to be a problem not depending on the kind of channel...happens with ISDN and PRI lines. What is stopping the frames, making Asterisk exiting that loop and dropping the calls? Thank you. Giorgio. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP
Michael Graves [EMAIL PROTECTED] writes: No, TCP for media as well. I though that was the whole point of SIP over TCP. Hopefully not. RTP over TCP would be entirely pointless. RTP needs packetization, doesn't mind packet loss (within reason) but hates retransmissions. TCP doesn't provide packetization, guarantees against packet loss, but retransmits. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building a Complex IVR
On Mon, 2008-06-23 at 09:54 -0700, Douglas Garstang wrote: I'm about to build a complex IVR with Asterisk. Having done it a few times with the dial plan, I know it's going to be pretty ugly. What are my other options? I guess I could do it in AGI/FastAGI. What about VxML (about which I know almost nothing...)? Using Asterisk 1.2 Thanks, Doug. Sorry, I tried to peak thru all the stuff in this thread, but I may have missed it; has anyone suggested the externalIVR app? If not, it might be worth consideration...? murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can I use X-Lite from local and external ip (when I'm not at home) ?
Hi, X-Lite demo version has only one SIP account possible. I'd like to set it up in such manner that I could register with Asterisk being at home (local LAN, local ip) and at work (external ip). Is this possible since X-Lite allows only one sip account settings ? Thanks in advance, regards, Rob. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21 + Realtime Queues = Agents Not Ringing?
Ex Vito wrote: I recently observed a similar behaviour under 1.4.21. The member was a SIP phone which had its calls forwarded to another SIP phone via its built-in configuration... (fyi: linksys spa922) For some reason, asterisk could not manage this scenario. I still have to test it better to understand if this is supposed to work or not. Could that be your case ? (not very probable, I know...) -- exvito No this is not my scenario, but thanks for checking. :) I am now building a virtual machine to test the 1.4 branch and figure out the issue, I'll forward anything I find. Unfortunately, I've not seen anyone else with this issue, as I've googled like crazy -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can asterisk support using different ip for rtp?
Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows RTP to use different IP as SIP ip. Is there any way to configure it? GUI or CLI? or , will we support it in future? Thanks. -- Rgds, -- Rgds, Hans Yin Web: homeofhans.homeip.net Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Skype: hans_yin_vancouver ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird one way Audio situation
Well, I have new information if anyone can/want to help me... (Please read all the previous messages in this email) If I call a number that can't hear me at all (calling from inside my network using a Grandstream GXP-2000 phone through Asterisk) and then I put this call on hold for a second and then I take again the call, then the callee start hearing me, :s Any ideas??? Thanks in advance... -- Nacho Linux Counter #156439 On Tue, Jun 17, 2008 at 7:50 PM, Raúl Gómez C. [EMAIL PROTECTED] wrote: I've been playing around in order to find something new and I've found this: I have created an IVR for test purposes, then I've placed a call from my sip phone using one of my telco lines to another of my telco lines attached to the PBX, in this situation I'm using two FXO channels, one for the outgoing call and another for the incoming call. Then I have created an extension in this IVR in order to make an echo test and I've used MixMonitor() to record the audio of the test. When I dial this extension I never can hear my echoed voice, but when I listen to the recording the audio have a lot of artifacts and the busy and dial tone are almost inaudible, the same effect that happens when you play to almost identical audio files, so I can presume that it is the same audio wave but out of phase (meaning the echo is working, I think). I don't know if this can be happening because of the Hardware Echo Canceler on my Remora A400D. If I call the extension of the echo test directly from my SIP phone without using any telco line (SIP -- IP -- Asterisk) then the test works just fine. Another test I've made is, during a call with the one way audio problem, I have used the ZapBarge() application to hear what's happening on the Zap Channel (from another SIP phone on my network). In this case I heard the callee complaining that he/she can't hear anything and I can't hear the caller (which is on the same network of my phone). In this case the caller can hear the callee. I have grabbed the sip debug messages of this call from the asterisk CLI and is attached (compressed) to this email. Well, thanks again for any comment/response... -- Nacho Linux Counter #156439 On Tue, Jun 17, 2008 at 5:14 PM, Raúl Gómez C. [EMAIL PROTECTED] wrote: Hi Steve and the rest of the list, On Thu, Jun 12, 2008 at 7:11 AM, Steve Totaro [EMAIL PROTECTED] wrote: Is your Asterisk box dual homed? Firewalled? Any output from the CLI with verbose turned on, that might help? Turn on SIP debugging as well. Thanks, Steve T My Asterisk Server has two NIC with a channel bonding setup (Balance TLB) connected to the same switch, and it does not have any firewall rule. I'm attaching a file with the output of sip set debug on the CLI of a call in this situation. Although calls made with SIP phones have this strange behavior, when I place a call with an analog phone connected to a FXS port of the same TDM card (see below for full description) this does not happen. Thanks, any help will be really appreciated... -- Nacho Linux Counter #156439 On Thu, Jun 12, 2008 at 7:11 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Tue, Jun 10, 2008 at 1:40 PM, Raúl Gómez C. [EMAIL PROTECTED] wrote: Hi list, I'm having trouble with calls placed to the PSTN (through a TDM card), sometimes (a lot indeed) when I dial a number the callee party can't hear me at all. My setup is: Asterisk 1.4.20.1 Zaptel 1.4.11 libpri 1.4.4 Wanpipe 3.2.4 I have a Sangoma Remora Card A400D (2 FXS / 10 FXO), Grandstream GXP-2000 IP Phones, SuSE Linux Enterprise Server 10 (SP2) x86_64 with Kernel 2.4.16.60-0.23-smp I'm using the ulaw audio codec. There is no NAT between the Asterisk Server and the Phones (the phone and the server are in the same network segment). What can it be??? Thanks in advance for any help/comment... -- Raul Linux Counter #156439 Is your Asterisk box dual homed? Firewalled? Any output from the CLI with verbose turned on, that might help? Turn on SIP debugging as well. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Nextone using H323
The meaning of softswitch in the Nextone is that it will try and use whatever protocol the corresponding Ingress/Egress side is set for. That means, if you place/receive a call from/to Asterisk, and route to an endpoint doing SIP, the Nextone will expect Asterisk to speak SIP. If you place/receive a call to/from an endpoint using H.323, it will expect Asterisk to speak H.323. When you use softswitch, the Nextone is much more forgiving about what it passes through also, expecting that your softswitch on the far side will take care of the issues. If you only want to use H.323 with Asterisk, you should configure it as an H.323 gateway. Why are you trying to set softswitch? That is how all of our systems are configured with Asterisk and ooh323. Works very well and very stable. Chris Everton Goularth wrote: Hi people, Someone have already used asterisk with Nextone? I`m trying to use it, but there are some problems.. One of these are when we set up a connection between Nextone and Asterisk using H323, we use our asterisk server as a Softswitch in the Nextone configuration, so it doesn`t work. But, when we just change (in Nextone configuration) from Softswitch to Gateway, it work. Where is the difference? I`m using chan_ooh323 in my asterisk server. This is my ooh323.conf: - [general] ;Default - 1720 ;port=1720 bindaddr= IP_ADDRESS ;This parameter indicates whether channel driver should register with ;gatekeeper as a gateway or an endpoint. ;Default - no ;gateway=yes ;Whether asterisk should use fast-start and tunneling for H323 connections. ;Default - yes ;faststart=no ;h245tunneling=no ;H323-ID to be used for asterisk server ;Default - Asterisk PBX h323id=GW2 e164=1521# ;CallerID to use for calls ;Default - Same as h323id callerid=MediaXChange 1.0 ;Whether this asterisk server will use gatekeeper. ;Default - DISABLE ;gatekeeper = DISCOVER ;gatekeeper = a.b.c.d ;gatekeeper = 189.44.163.125 logfile=/var/log/asterisk/h323_log context=default disallow=all allow=g729 allow=g723 dtmfmode=rfc2833 - Anyone know what can I do? or what am I doing wrong??? Thank`s a lot for the opportunity. Everton Goularth ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue with different music for each caller
Hello Thomas, no problem. In asterisk 1.6 use SetMusicOnHold(musiconholdname) then it will work in older Asterisk versions! br, Martin - Original Message - From: Thomas Winter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 24, 2008 5:50 PM Subject: Re: [asterisk-users] Queue with different music for each caller On Tuesday 24 June 2008 15:22, Martin Schrott - thinking:systems wrote: Hello Thomas you can use different music for each caller if you like. in extensions.conf you can set the music class. exten = s,n,Set(CHANNEL(musicclass)=yourmusicforthiscaller) Hi Martin, thanks for your suggestion, I forgot to notice that Iam still using 1.2.X Jun 24 17:45:31 ERROR[17784]: pbx.c:1437 ast_func_write: Function CHANNEL not registered So, this didnt work for me. best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Major problem with 1.4.21 asterisk
Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having major iax2 problems. All of a sudden calls wouldnt come in on the iax2 DID, and we couldnt make calls out even though everything looked ok. Also there was usually a hung iax2 channel when this happened. Stopping asterisk also wouldnt work, i would do a Stop now and it would just go back to the cli prompt. I would do a ? and it wouldnt work. I would have to kill asterisk via ps and then restart it via init.d and then iax2 would start working again for a short while (maybe a few hours) I reinstalled 1.4.19 and the problems went away. There appears to be a major bug in 1.4.21 but i am not sure. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
