Re: [asterisk-users] Asterisk Web Meetme module not loading
On 1/09/09 5:53 PM, Glen wrote: Matt Riddell wrote: In the latest readme for WebMeetMe (3.1.0) it states: * Compile and install CBMySQL App_cbmysql is now included in the web-meetme package, located in ./cbmysql. To install just run make; make install Copy the sample cbmysql.conf to /etc/asterisk and create a dialplan similar to the one in cb-extensions.conf.sample Modify the settings to suit your system. The location of the mysql.sock file is likely not correct, check /etc/my.conf for the correct location. That fixed it Matt, just compiling in the wrong directory. Thanks for all your help. No problems :) I haven't actually used it myself, but it looks pretty cool! -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] jitterbuffer for chan_sip on asterisk 1.2
Hello, From http://www.voip-info.org/wiki/view/Asterisk+new+jitterbuffer, it says that there For Asterisk 1.2 there was no jitterbuffer in the RTP-based channels (i.e. chan_sip). I am using 1.2 and Ind there is no reason to upgrade. Are there any developments on this? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Inquiry:Problem with VoiceMail
Dear All
Can you please do me favor and let me know what is my problem with my
Asterisk VoiceMail configuration as it doesn't work correctly in my case ?
Please find below that part of my extensions.conf that I intend to make use
of voice mail for No Answer reply :
[line-incoming]
exten = _XXX,1,macro(dialuser,SIP/${EXTEN},${EXTEN})
[macro-dialuser]
exten = s,1,dial(${ARG1},38,r)
exten = s,n,noop(PM: Dial ended !!)
exten = s,n,noop(${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,answer
exten = s-NOANSWER,n,wait(4)
exten = s-NOANSWER,n,SayDigits(${ARG2})
exten = s-NOANSWER,n,playback(vm-isunavail)
exten = s-NOANSWER,n,VoiceMail(u${MACRO_EXTEN})
exten = s-NOANSWER,n,hangup()
As you see , I intend to redirect the calling party to the called party
voice mailbox if he doesn't answer the call (that will be set at the number
the same as his extension number) but it doesn't get through. Can you please
let me know what is wrong in our case ?
Regards
H.Motamedi
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Re: [asterisk-users] CDR to Postgres Centos
Thanks alot Barry this was really helpful Can i know which querry is executed to insert record to database... i am asking this because of [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:309 pgsql_log: Failed to insert call detail record into database! [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:310 pgsql_log: Reason: ERROR: syntax error at or near ) at character 17 [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:311 pgsql_log: Connection may have been lost... attempting to reconnect. [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:314 pgsql_log: Connection reestablished. [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:320 pgsql_log: HARD ERROR! Attempted reconnection failed. DROPPING CALL RECORD! On Mon, Aug 31, 2009 at 6:06 PM, Barry L. Kline blkl...@attglobal.netwrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 ABBAS SHAKEEL wrote: but when i execute this ./configure --with-postgres=dir where postgresql is installed it gives an error for missing an pg_config file . i searched the PC but it really dont exists. but database server is fine and working OK! what to do in this situation You should have the following packages installed on your Asterisk system postgresql-libs postgresql-devel postgresql If the database is on the same box, include: postgresql-server If you want to hit the database from the dialplan for any reason, include: postgresql-odbc Once you install these, be sure to rerun ./configure Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKm8rOCFu3bIiwtTARAijbAJ4vt0DVZJYUPRhPrNpXm2KEngRmxACgn24T aHtpBzyGhPBmw8a4veqdLhQ= =TI+m -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] SIP and other phones other then local network
Hello Please advice how can i configure a sip phone that is not on my local network. ie i have Xlite far some where in America and my Asterisk server is at Sahara desert . Now how can i make a call to that sip phone? Please advice what keywords to carry on?? -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Inquiry:Problem with VoiceMail
On 1/09/09 6:14 PM, hadi motamedi wrote:
exten = s,n,noop(${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1)
As you see , I intend to redirect the calling party to the called party
voice mailbox if he doesn't answer the call (that will be set at the
number the same as his extension number) but it doesn't get through. Can
you please let me know what is wrong in our case ?
In the Asterisk console, what does it say the dialstatus is?
--
Cheers,
Matt Riddell
Director
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http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
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Re: [asterisk-users] SIP and other phones other then local network
On 1/09/09 7:48 PM, ABBAS SHAKEEL wrote: Hello Please advice how can i configure a sip phone that is not on my local network. ie i have Xlite far some where in America and my Asterisk server is at Sahara desert . Now how can i make a call to that sip phone? Please advice what keywords to carry on?? Search for Asterisk SIP NAT. Basically you'll need to port forward 5060 and the rtp ports (1-2 by default) to your Asterisk machine from the firewall. The outside person then registers to your machine (by using the external address). -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Inquiry:Problem with VoiceMail
Thank you for your reply . The part of the code that I sent you is for the
case the called party didn't answer . I want to get the calling party
message into the called party voice mail box . Please help me to correct my
code .
Regards
H.Motamedi
On Tue, Sep 1, 2009 at 8:52 AM, Matt Riddell li...@venturevoip.com wrote:
On 1/09/09 6:14 PM, hadi motamedi wrote:
exten = s,n,noop(${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1)
As you see , I intend to redirect the calling party to the called party
voice mailbox if he doesn't answer the call (that will be set at the
number the same as his extension number) but it doesn't get through. Can
you please let me know what is wrong in our case ?
In the Asterisk console, what does it say the dialstatus is?
--
Cheers,
Matt Riddell
Director
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http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
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Re: [asterisk-users] Inquiry:Problem with VoiceMail
On 1/09/09 8:45 PM, hadi motamedi wrote: Thank you for your reply . The part of the code that I sent you is for the case the called party didn't answer . I want to get the calling party message into the called party voice mail box . Please help me to correct my code . If you answered my question I could help you! :) You know you have an Asterisk console? Like when you type asterisk -r Ok, so let's go into the Asterisk console. Once you're there type: core set verbose 3 Then make a phone call to someone who will not answer (so you can see what happens). When this fails, use your mouse to select the text that you see in the console, copy it, and paste it into an email. If you can't use your mouse (because you are actually sitting at the Asterisk server, not logged in) then you might like to use some software like putty from your desktop machine. Once you have the info pasted into an email, send it to this list so we can read it and see what is wrong. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] jitterbuffer for chan_sip on asterisk 1.2
1 sep 2009 kl. 08.17 skrev James Mutuku: Hello, From http://www.voip-info.org/wiki/view/Asterisk+new+jitterbuffer, it says that there For Asterisk 1.2 there was no jitterbuffer in the RTP-based channels (i.e. chan_sip). I am using 1.2 and Ind there is no reason to upgrade. Are there any developments on this? Well, the development ended up being named Asterisk 1.4 which included jitter buffers. That's a good reason to update! /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Selective canreinvite in multi-tenant environment
1 sep 2009 kl. 05.18 skrev John A. Sullivan III: On Thu, 2009-08-27 at 14:23 -0400, John A. Sullivan III wrote: Hello, all. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, we can't, but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic associated with intra-tenant traffic from the Asterisk server and reduce the intra-tenant latency by doing so. However, I am very, very hesitant to allow our VPN connections to tenants to function as a router between tenants allowing one tenant to directly access phones on another tenant (that's not as wild as it sounds because of our use of the ISCS project - iscs.sourceforge.net). Since the tenants are all connecting via VPN, we are using RFC1918 addresses and no NAT is involved thus the canreinvite=nonat option does not help us. If we set canreinvite=nonat, that will allow for intra-tenant direct media but, if one tenant tries to call another via SIP, it will redirect the media at the Asterisk level but the packets will be dropped at the firewall / router level (or sooner as there may be no route to the destination) and the call will connect but with no sound. Any guidance would be greatly appreciated. Thanks - John As mentioned in another post, we were able to solve this by setting a w dial option to all inbound SIP calls from the Internet. Thus, all internal calls could reinvite but external calls could not. However, just when we thought this was working splendidly well, we turned up another roadblock - transfers. We find that once we transfer a call using our Snom phones, the call between the new call partners does not seem bound by the w constraint, Asterisk tries to reinvite the call, and the audio breaks because the firewall cannot associate the new RTP stream with a SIP session. How have others gotten around the problem of transfers causing reinvites on calls which should not allow reinvites? Thanks - John I think this is an issue that needs some code to solve it, so you can set a variable in the dialplan that prevents remote RTP bridges (reinvited media). Contact me off list if you're interested in sponsoring such development. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Congratulations to Kamailio - Infoworld Best of Open Source Awards
Friends, I would like to congratulate kamailio.org - a project we're cooperating a lot with. They have just been awarded the BOSSIE award by InfoWorld. Kamailio is the OpenSER SIP proxy project with a new name, a product widely used in Asterisk installations. And of course, the motivation mentions Asterisk :-) From InfoWorld site: Award winners in network and network management are old favorites Cacti and Nagios, the IPCop firewall, Kamailio SIP proxy server, KeePass password manager, Openfiler SAN/NAS appliance, OpenNMS enterprise monitoring system, PacketFence network access control solution, Puppet configuration management framework, and Untangle network security gateway. Kamailio is the open source SIP proxy server formerly known as OpenSER. Used with an Asterisk IP PBX server for phone features, plus a hardware gateway for connection to the outside world, Kamailio brings important call handling and scalability benefits to Asterisk, while also removing the Asterisk server as a single point of failure. Larger organizations get the phone features they need, as well as the added safety of VoIP calls surviving an Asterisk server outage. http://infoworld.com/d/open-source/best-open-source-software-awards-2009-628 Congratulations to the whole Kamailio.org development team! Regards, /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] jitterbuffer for chan_sip on asterisk 1.2
The project I am working on is really big. Unless I upgrade during christmas(by then the project will be several months overdue). Just googled further and saw some patches. I will try them and see. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] jitterbuffer for chan_sip on asterisk 1.2
On 1/09/09 9:43 PM, James Mutuku wrote: The project I am working on is really big. Unless I upgrade during christmas(by then the project will be several months overdue). Just googled further and saw some patches. I will try them and see. In which case you probably shouldn't be using Asterisk 1.2 -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] jitterbuffer for chan_sip on asterisk 1.2
I did am not the one who started the project. the client has been running 1.2 for years and they needed additional features set up ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 issue
On Tue, Sep 1, 2009 at 4:35 AM, Paul Halespdha...@optusnet.com.au wrote: Miguel Molina wrote: Paul Hales escribió: I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH Hi, Maybe maxlen = 1? Cheers, Hmmm - almost. Maxlen limits the amounts of calls waiting for the queue, not the amount of callers talking to queue members. You can do any limitations i can imagine with Set(GROUP()=...) and GROUP_COUNT. Do You actually need rest of callers to wait in queue while one is speaking, or disconnect them before they enter queue? Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] jitterbuffer for chan_sip on asterisk 1.2
On 1/09/09 10:02 PM, James Mutuku wrote: I did am not the one who started the project. the client has been running 1.2 for years and they needed additional features set up There was an Asterisk backports site - you might want to check in google -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Multiple user registration ...
Thanks a lot Faheem for you help.
I totaly understand now the approach you've used.
It's very interesting and inventive for sure.
I didn't know that I could append IP:Port info on user when using the
Dial command and that this will make calling to two different devices
registered using same user work.
With this little but extemelly important peace of information you gave
me the answer to our questions here.
Thanks again, and best regards,
Mauro.
Faheem escreveu:
The purpose of Perl script is to store user registrations records only
and nothing else regarding call dialing.
The script will main records like this.
User1:
IP1: 192.168.0.100 Por1: 5060
IP2: 69.30.21.10 Port2: 5060
User2:
IP1: 192.168.10.1 Por1: 5060
IP2: 192.168.10.1 Por2: 5061
User3:
IP1: 192.168.10.121 Por1: 5060
IP2: 192.168.10.123 Por2: 5061
and so on
No it all depends on you to store these information on files or database.
Assume you have stored IP/Ports in the database.
Database=cloneline
Table = users(username,ip1,port1,ip2,port2)
For dialing:
Assume username=user1 and extension =123456
exten= 123456,1,NoOp()
exten= 123456,n,MYSQL(Connect connid 'localhost' cdr dbpass cloneline)
exten= 123456,n,NoOP(Connection ID:${connid})
exten= 123456,n,MYSQL(Query resultid ${connid} SELECT\ ip1\, port1\,
ip2\, port2\, status\ from\ users\ where\ username=user1 )
exten= 123456,n,MYSQL(Fetch fetchid ${resultid} ip1 port1 ip2 port2)
exten= 123456,n,Dial(SIP/us...@${ip1}:${port1}SIP/us...@${ip2}:${port2})
for dialing user3
username=user3 and extension =112233
exten= 112233,1,NoOp()
exten= 112233,n,MYSQL(Connect connid 'localhost' cdr dbpass cloneline)
exten= 112233,n,NoOP(Connection ID:${connid})
exten= 112233,n,MYSQL(Query resultid ${connid} SELECT\ ip1\, port1\,
ip2\, port2\, status\ from\ users\ where\ username=user3 )
exten= 112233,n,MYSQL(Fetch fetchid ${resultid} ip1 port1 ip2 port2)
exten= 112233,n,Dial(SIP/us...@${ip1}:${port1}SIP/us...@${ip2}:${port2})
Hope every thing would be clear...
Muhammad Faheem
Software Engineer
AxVoice Inc.
307,Y Commercial,
DHA Lahore, Pakistan
+92-333-4793314
http://www.axvoice.com
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__At.,
_
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br mailto:@tqi.com.br
: www.tqi.com.br http://www.tqi.com.br
( + 55 (34)3291-1700
( + 55 (34)9971-2572
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Re: [asterisk-users] CDR to Postgres Centos
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 ABBAS SHAKEEL wrote: Can i know which querry is executed to insert record to database... i am asking this because of [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:309 pgsql_log: Failed to insert call detail record into database! [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:310 pgsql_log: Reason: ERROR: syntax error at or near ) at character 17 [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:311 pgsql_log: Connection may have been lost... attempting to reconnect. [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:314 pgsql_log: Connection reestablished. [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:320 pgsql_log: HARD ERROR! Attempted reconnection failed. DROPPING CALL RECORD! Which version of Asterisk are you using? Did you create the PG database for Asterisk? Have you confirmed that you can connect to it using the CLI psql with the appropriate credentials? There are a few steps ahead of where you are before we worry about this particular problem. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKnRL4CFu3bIiwtTARAnwNAJ9+CiWdtq17DRSqelNl7bsN5pS32gCeIn+l VNyWYBauMOBvVMhyGUeP/Pk= =G9NP -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Dahdi configuraion / error
Hello I just updated the kernel, dahdi-linux and dahdi-tools Im also using now asterisk 1.4.26.1 And im still with a red light (not RED/YELLOW anymore): [r...@catumbela ~]# /etc/rc.d/init.d/dahdi status ### Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED 1 PRICAS RED 2 PRICAS RED 3 PRICAS RED 4 PRICAS RED 5 PRICAS RED 6 PRICAS RED 7 PRICAS RED 8 PRICAS RED 9 PRICAS RED 10 PRICAS RED 11 PRICAS RED 12 PRICAS RED 13 PRICAS RED 14 PRICAS RED 15 PRICAS RED 16 PRIHDLCFCS RED 17 PRICAS RED 18 PRICAS RED 19 PRICAS RED 20 PRICAS RED 21 PRICAS RED 22 PRICAS RED 23 PRICAS RED 24 PRICAS RED 25 PRICAS RED 26 PRICAS RED 27 PRICAS RED 28 PRICAS RED 29 PRICAS RED 30 PRICAS RED 31 PRICAS RED Here is my config: /etc/dahdi/system.conf -- loadzone = us defaultzone=us span=1,1,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101 tail -f /var/log/messages Sep 1 13:56:08 catumbela kernel: dahdi: Telephony Interface Registered on major 196 Sep 1 13:56:08 catumbela kernel: dahdi: Version: 2.2.0.2 Sep 1 13:56:08 catumbela kernel: Found TE4XXP at base address fdcff000, remapped to f88a8000 Sep 1 13:56:08 catumbela kernel: TE4XXP version c01a0164, burst OFF Sep 1 13:56:08 catumbela kernel: FALC version: 0005, Board ID: 00 Sep 1 13:56:08 catumbela kernel: Reg 0: 0x35dbc400 Sep 1 13:56:08 catumbela kernel: Reg 1: 0x35dbc000 Sep 1 13:56:08 catumbela kernel: Reg 2: 0x Sep 1 13:56:08 catumbela kernel: Reg 3: 0x Sep 1 13:56:08 catumbela kernel: Reg 4: 0x Sep 1 13:56:08 catumbela kernel: Reg 5: 0x Sep 1 13:56:08 catumbela kernel: Reg 6: 0xc01a0164 Sep 1 13:56:08 catumbela kernel: Reg 7: 0x1f00 Sep 1 13:56:08 catumbela kernel: Reg 8: 0x010200ff Sep 1 13:56:08 catumbela kernel: Reg 9: 0x00fd Sep 1 13:56:08 catumbela kernel: Reg 10: 0x004a Sep 1 13:56:08 catumbela kernel: Found a Wildcard: Wildcard TE405P (2nd Gen) Sep 1 13:56:08 catumbela kernel: TE4XXP: Launching card: 0 Sep 1 13:56:08 catumbela kernel: TE4XXP: Setting up global serial parameters Sep 1 13:56:08 catumbela dahdi: wct4xxp: succeeded Sep 1 13:56:14 catumbela kernel: About to enter spanconfig! Sep 1 13:56:14 catumbela kernel: Done with spanconfig! Sep 1 13:56:14 catumbela kernel: dahdi: Registered tone zone 0 (United States / North America) Sep 1 13:56:14 catumbela kernel: About to enter startup! Sep 1 13:56:14 catumbela kernel: TE4XXP: Span 1 configured for CAS/HDB3 Sep 1 13:56:14 catumbela kernel: wct4xxp: Setting yellow alarm on span 1 Sep 1 13:56:14 catumbela kernel: timing source auto card 0! Sep 1 13:56:14 catumbela kernel: SPAN 1: Primary Sync Source Sep 1 13:56:14 catumbela kernel: VPM400: Not Present Sep 1 13:56:14 catumbela kernel: VPM450: Not Present Sep 1 13:56:14 catumbela kernel: Completed startup! Sep 1 13:56:14 catumbela dahdi: Running dahdi_cfg: succeeded here is dmesg: DMESG - [r...@catumbela ~]# dmesg Linux version 2.6.9-89.0.9.ELsmp (mockbu...@builder10.centos.org) (gcc version 3.4.6 20060404 (Red Hat 3.4.6-11)) #1 SMP Mon Aug 24 07:56:18 EDT 2009 BIOS-provided physical RAM map: BIOS-e820: - 0009c000 (usable) BIOS-e820: 0009c000 - 000a (reserved) BIOS-e820: 000f - 0010 (reserved) BIOS-e820: 0010 - 7bf0 (usable) BIOS-e820: 7c00 - 8000 (reserved) BIOS-e820: e000 - f000 (reserved) BIOS-e820: fec0 - 0001 (reserved) 1087MB HIGHMEM available. 896MB LOWMEM available. found SMP MP-table at 000f3ab0 NX (Execute Disable) protection: active On node 0 totalpages: 507648 DMA zone: 4096 pages, LIFO batch:1 Normal zone: 225280 pages, LIFO batch:16 HighMem zone: 278272 pages, LIFO batch:16 DMI 2.2 present. Using APIC driver default ACPI: Unable to locate RSDP Nvidia board detected. Ignoring ACPI timer override. Intel MultiProcessor Specification v1.4 Virtual Wire compatibility mode. OEM ID: OEM0 Product ID: PROD APIC at: 0xFEE0 Processor #0 15:3 APIC version 17 Processor #1 15:3 APIC version 17 I/O APIC #2 Version 17 at 0xFEC0. Enabling APIC mode: Flat. Using 1 I/O APICs Processors: 2 Allocating PCI resources starting at 8800 (gap: 8000:6000) Built 1 zonelists Kernel command
Re: [asterisk-users] Dahdi configuraion / error
This may be dumb and/or obvious, but did you do these steps? 1. dahdi_genconf dahdi modules user to make sure all of the configuration files are up to standard 2. dahdi_cfg -vv to see if any obvious messages came up (causing the red condition(s) ) The information you have provided is useful, but it boils down to this (IMO) - RED is dead! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes Pereira Sent: Tuesday, September 01, 2009 8:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dahdi configuraion / error Hello I just updated the kernel, dahdi-linux and dahdi-tools Im also using now asterisk 1.4.26.1 And im still with a red light (not RED/YELLOW anymore): [r...@catumbela ~]# /etc/rc.d/init.d/dahdi status ### Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED 1 PRICAS RED 2 PRICAS RED 3 PRICAS RED 4 PRICAS RED 5 PRICAS RED 6 PRICAS RED 7 PRICAS RED 8 PRICAS RED 9 PRICAS RED 10 PRICAS RED 11 PRICAS RED 12 PRICAS RED 13 PRICAS RED 14 PRICAS RED 15 PRICAS RED 16 PRIHDLCFCS RED 17 PRICAS RED 18 PRICAS RED 19 PRICAS RED 20 PRICAS RED 21 PRICAS RED 22 PRICAS RED 23 PRICAS RED 24 PRICAS RED 25 PRICAS RED 26 PRICAS RED 27 PRICAS RED 28 PRICAS RED 29 PRICAS RED 30 PRICAS RED 31 PRICAS RED Here is my config: /etc/dahdi/system.conf -- loadzone = us defaultzone=us span=1,1,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101 tail -f /var/log/messages Sep 1 13:56:08 catumbela kernel: dahdi: Telephony Interface Registered on major 196 Sep 1 13:56:08 catumbela kernel: dahdi: Version: 2.2.0.2 Sep 1 13:56:08 catumbela kernel: Found TE4XXP at base address fdcff000, remapped to f88a8000 Sep 1 13:56:08 catumbela kernel: TE4XXP version c01a0164, burst OFF Sep 1 13:56:08 catumbela kernel: FALC version: 0005, Board ID: 00 Sep 1 13:56:08 catumbela kernel: Reg 0: 0x35dbc400 Sep 1 13:56:08 catumbela kernel: Reg 1: 0x35dbc000 Sep 1 13:56:08 catumbela kernel: Reg 2: 0x Sep 1 13:56:08 catumbela kernel: Reg 3: 0x Sep 1 13:56:08 catumbela kernel: Reg 4: 0x Sep 1 13:56:08 catumbela kernel: Reg 5: 0x Sep 1 13:56:08 catumbela kernel: Reg 6: 0xc01a0164 Sep 1 13:56:08 catumbela kernel: Reg 7: 0x1f00 Sep 1 13:56:08 catumbela kernel: Reg 8: 0x010200ff Sep 1 13:56:08 catumbela kernel: Reg 9: 0x00fd Sep 1 13:56:08 catumbela kernel: Reg 10: 0x004a Sep 1 13:56:08 catumbela kernel: Found a Wildcard: Wildcard TE405P (2nd Gen) Sep 1 13:56:08 catumbela kernel: TE4XXP: Launching card: 0 Sep 1 13:56:08 catumbela kernel: TE4XXP: Setting up global serial parameters Sep 1 13:56:08 catumbela dahdi: wct4xxp: succeeded Sep 1 13:56:14 catumbela kernel: About to enter spanconfig! Sep 1 13:56:14 catumbela kernel: Done with spanconfig! Sep 1 13:56:14 catumbela kernel: dahdi: Registered tone zone 0 (United States / North America) Sep 1 13:56:14 catumbela kernel: About to enter startup! Sep 1 13:56:14 catumbela kernel: TE4XXP: Span 1 configured for CAS/HDB3 Sep 1 13:56:14 catumbela kernel: wct4xxp: Setting yellow alarm on span 1 Sep 1 13:56:14 catumbela kernel: timing source auto card 0! Sep 1 13:56:14 catumbela kernel: SPAN 1: Primary Sync Source Sep 1 13:56:14 catumbela kernel: VPM400: Not Present Sep 1 13:56:14 catumbela kernel: VPM450: Not Present Sep 1 13:56:14 catumbela kernel: Completed startup! Sep 1 13:56:14 catumbela dahdi: Running dahdi_cfg: succeeded here is dmesg: DMESG - [r...@catumbela ~]# dmesg Linux version 2.6.9-89.0.9.ELsmp (mockbu...@builder10.centos.org) (gcc version 3.4.6 20060404 (Red Hat 3.4.6-11)) #1 SMP Mon Aug 24 07:56:18 EDT 2009 BIOS-provided physical RAM map: BIOS-e820: - 0009c000 (usable) BIOS-e820: 0009c000 - 000a (reserved) BIOS-e820: 000f - 0010 (reserved) BIOS-e820: 0010 - 7bf0 (usable) BIOS-e820: 7c00 - 8000 (reserved) BIOS-e820: e000 - f000 (reserved) BIOS-e820: fec0 - 0001 (reserved) 1087MB HIGHMEM available. 896MB LOWMEM available. found SMP MP-table at 000f3ab0 NX (Execute Disable) protection: active On node 0 totalpages: 507648 DMA zone:
[asterisk-users] set language in asterisk-1.6.x
Hello, How can Set(language()) in asterisk-1.6.x ? Cordialement, P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] List Access
On Aug 31, 2009, at 3:01 PM, David @ULC wrote: To view the post and reply , I always to use below link.. http://lists.digium.com/pipermail/asterisk-users/2009-August/thread.html Any better way to access the forum ? There are a variety of other locations that subscribe to the asterisk- users mailing list and archive in their own formats - some quick Google searching will find them, but honestly I don't keep a list since I use the canonical source, which is lists.digium.com. You may want to use Gmail and create your own archive - I know quite a few people like Gmail's threading capabilities for list reading/ archiving. Just get a Gmail account and then filter your asterisk-* into a separate folder. JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Dahdi configuraion / error
On Tue, Sep 01, 2009 at 08:53:04AM -0500, Danny Nicholas wrote: This may be dumb and/or obvious, but did you do these steps? 1. dahdi_genconf dahdi modules user to make sure all of the configuration files are up to standard (Which will default to generate a ccs configuration for it, rather than cas) Current configuration appears to be OK at first glance. 2. dahdi_cfg -vv to see if any obvious messages came up (causing the red condition(s) ) The information you have provided is useful, but it boils down to this (IMO) - RED is dead! Is there actually a cable plugged? Connecting it to a live system? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] set language in asterisk-1.6.x
Sorry, I just found the solution Set(CHANNEL(language)=en) Cordialement, P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de BERGANZ François Envoyé : mardi 1 septembre 2009 16:15 À : asterisk-users@lists.digium.com Objet : [asterisk-users] set language in asterisk-1.6.x Hello, How can Set(language()) in asterisk-1.6.x ? Cordialement, P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 and other phones other then local network
For example if it was Alex to reply to that msg, i would feel bad for this guy, because Alex would make him feel like if he cannot do this by himself or use google to find that answer by himself, he does not belong to that list. He would never give him a chance and try to help him. Sent from my iPod On Sep 1, 2009, at 3:53 AM, Matt Riddell li...@venturevoip.com wrote: On 1/09/09 7:48 PM, ABBAS SHAKEEL wrote: Hello Please advice how can i configure a sip phone that is not on my local network. ie i have Xlite far some where in America and my Asterisk server is at Sahara desert . Now how can i make a call to that sip phone? Please advice what keywords to carry on?? Search for Asterisk SIP NAT. Basically you'll need to port forward 5060 and the rtp ports (1-2 by default) to your Asterisk machine from the firewall. The outside person then registers to your machine (by using the external address). -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] Digium PRI cards for data usage?
- Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Monday 31 August 2009 21:59:28 Tim Nelson wrote: Greetings- I'm wondering if the Digium PRI cards can be used for data (Cisco HDLC, PPP, etc) or if they're for voice circuits only. I haven't been able to find any information on this. All documentation direct from Digium seems to indicate their hardware is for voice applications only. Sangoma's cards work in either voice or data mode but of course this is configured in their Wanpipe software. Thanks for any pointers. You can. The keyword is nethdlc in /etc/dahdi/system.conf, although to enable it, you need to uncomment CONFIG_DAHDI_NET in include/dahdi/dahdi_config.h and recompile the dahdi drivers. Once the active spans are configured with nethdlc, use the sethdlc command line utility to set up the bonded channels into the various network interfaces (hdlc0 through hdlcN). Depending upon your configuration, you may or may not also need to then configure the corresponding pvcN devices. Here is an article on the old Zaptel interface. While the name of the driver may have changed, the procedures remain the same: http://www.softwink.com/papers/Installation_Securing_VoIP_With_Linux/ By the way, the method for determining which channels are bonded are as simple as the number of channels you configure together (on a single line) in /etc/dahdi/system.conf. For example, you can do as little as nethdlc=1 (for a single 64k channel) up to nethdlc=1-192 (for 8 T1s bonded into a single data device). Each nethdlc line in the config becomes a separate hdlcN device. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) Thank you Tilghman! That is exactly what I've been looking for! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] List Access
On Mon, Aug 31, 2009 at 2:01 PM, David @ULCucoms2...@gmail.com wrote: Any better way to access the forum ? I think 'better' is pretty subjective, but I find reading the list with GNUS using the gmane.org NNTP service pretty easy going. I'm also subscribed to the list with my gmail address, and often post from here. -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Digium PRI cards for data usage?
Do you have to set aside kines for the data channels or can you have dynamic data channels, for example ISDN dialup internet backup? Robb 2009/9/1 Tim Nelson tnel...@rockbochs.com - Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Monday 31 August 2009 21:59:28 Tim Nelson wrote: Greetings- I'm wondering if the Digium PRI cards can be used for data (Cisco HDLC, PPP, etc) or if they're for voice circuits only. I haven't been able to find any information on this. All documentation direct from Digium seems to indicate their hardware is for voice applications only. Sangoma's cards work in either voice or data mode but of course this is configured in their Wanpipe software. Thanks for any pointers. You can. The keyword is nethdlc in /etc/dahdi/system.conf, although to enable it, you need to uncomment CONFIG_DAHDI_NET in include/dahdi/dahdi_config.h and recompile the dahdi drivers. Once the active spans are configured with nethdlc, use the sethdlc command line utility to set up the bonded channels into the various network interfaces (hdlc0 through hdlcN). Depending upon your configuration, you may or may not also need to then configure the corresponding pvcN devices. Here is an article on the old Zaptel interface. While the name of the driver may have changed, the procedures remain the same: http://www.softwink.com/papers/Installation_Securing_VoIP_With_Linux/ By the way, the method for determining which channels are bonded are as simple as the number of channels you configure together (on a single line) in /etc/dahdi/system.conf. For example, you can do as little as nethdlc=1 (for a single 64k channel) up to nethdlc=1-192 (for 8 T1s bonded into a single data device). Each nethdlc line in the config becomes a separate hdlcN device. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) Thank you Tilghman! That is exactly what I've been looking for! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] jitterbuffer for chan_sip on asterisk 1.2
Matt Riddell wrote: On 1/09/09 10:02 PM, James Mutuku wrote: I did am not the one who started the project. the client has been running 1.2 for years and they needed additional features set up There was an Asterisk backports site - you might want to check in google Pretty sure that site is long gone -- I searched for it a while ago, and it just comes up with a parked domain image. Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] jitterbuffer for chan_sip on asterisk 1.2
I doubt using one of the patches is a good idea either, it will lack the needed testing and it's all quite fragile. Zoa Leif Madsen wrote: Matt Riddell wrote: On 1/09/09 10:02 PM, James Mutuku wrote: I did am not the one who started the project. the client has been running 1.2 for years and they needed additional features set up There was an Asterisk backports site - you might want to check in google Pretty sure that site is long gone -- I searched for it a while ago, and it just comes up with a parked domain image. Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] jitterbuffer for chan_sip on asterisk 1.2
It's long gone. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 MWI issue
Hi, I'd appreciate any comments please. Thanks. - Original Message - From: ilker Aktuna To: asterisk-users@lists.digium.com Sent: Monday, August 31, 2009 8:11 PM Subject: [asterisk-users] Asterisk MWI issue Hi, I am using Asterisk personally at home. My SIP client (SPA 3000) supports MWI with SIP NOTIFY messages. With a previous version of Asterisk I had no problems with MWI. But now I am using the following version which comes with Trixbox 2.8.0.1, and I have problems with MWI. Asterisk 1.6.0.9-samy-r27 Problem description: When a voicemail is left on the extension, a SIP NOTIFY message is sent to my SIP client and the MWI is received ok. This is good. But when I delete all Voicemail through AMPortal, SIP NOTIFY message notifying that there is no voicemail left is not sent to the client. Normally , I expect to receie a SIP NOTIFY message as soon as my inbox is empty. However, this does not happen when I delete voicemail through GUI. If I delete voicemail through phone, I receive the SIP NOTIFY message. How can I fix this problem ? Is this a misconfiguration or a bug ? If it is a bug, is there any fix for it ? Thanks, ilker -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 MWI issue
On Tue, 1 Sep 2009, ilker Aktuna wrote: I'd appreciate any comments please. You've been asking this question for several days now. It's been suggested that a Trixbox forum would be more productive. If this was an Asterisk issue, it is probable that a non-Trixbox user would also have discovered it and it would have been discussed. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 MWI issue
It's (IMO) a TRIXBOX bug. To verify this, leave yourself a voicemail, then delete it through Asterisk. If the behavior is as you expect, then you need to see what gets zapped in Asterisk that TB does not Zap. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ilker Aktuna Sent: Tuesday, September 01, 2009 11:56 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk MWI issue Hi, I'd appreciate any comments please. Thanks. - Original Message - From: ilker Aktuna mailto:ilk...@kobiline.com To: asterisk-users@lists.digium.com Sent: Monday, August 31, 2009 8:11 PM Subject: [asterisk-users] Asterisk MWI issue Hi, I am using Asterisk personally at home. My SIP client (SPA 3000) supports MWI with SIP NOTIFY messages. With a previous version of Asterisk I had no problems with MWI. But now I am using the following version which comes with Trixbox 2.8.0.1, and I have problems with MWI. Asterisk 1.6.0.9-samy-r27 Problem description: When a voicemail is left on the extension, a SIP NOTIFY message is sent to my SIP client and the MWI is received ok. This is good. But when I delete all Voicemail through AMPortal, SIP NOTIFY message notifying that there is no voicemail left is not sent to the client. Normally , I expect to receie a SIP NOTIFY message as soon as my inbox is empty. However, this does not happen when I delete voicemail through GUI. If I delete voicemail through phone, I receive the SIP NOTIFY message. How can I fix this problem ? Is this a misconfiguration or a bug ? If it is a bug, is there any fix for it ? Thanks, ilker _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 MWI issue
- Original Message - From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 01, 2009 8:20 PM Subject: Re: [asterisk-users] Asterisk MWI issue On Tue, 1 Sep 2009, ilker Aktuna wrote: I'd appreciate any comments please. You've been asking this question for several days now. It's been suggested that a Trixbox forum would be more productive. If this was an Asterisk issue, it is probable that a non-Trixbox user would also have discovered it and it would have been discussed. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Hi Steve, I understand your point but I've already tried my luck on Trixbox forums. Nothing came out from there. I don't understand what you mean by trixbox or Asterisk issue Trixbox uses Asterisk as the core. It does not do anything special... Maybe Trixbox has a special Asterisk build and updating it would save me. But I can't be sure. I just need help (other than pointing other forums that are really useless) Thanks, ilker ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 MWI issue
Hi Danny, Thanks for your suggestion. How can I delete voicemail through Asterisk (without using Trixbox GUI which is in fact AMPortal) ? Thanks, ilker - Original Message - From: Danny Nicholas To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, September 01, 2009 8:28 PM Subject: Re: [asterisk-users] Asterisk MWI issue It's (IMO) a TRIXBOX bug. To verify this, leave yourself a voicemail, then delete it through Asterisk. If the behavior is as you expect, then you need to see what gets zapped in Asterisk that TB does not Zap. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 MWI issue
On Mon, 31 Aug 2009, ilker Aktuna wrote: When a voicemail is left on the extension, a SIP NOTIFY message is Sent to my SIP client and the MWI is received ok. This is good. But when I delete all Voicemail through AMPortal, SIP NOTIFY message notifying that there is no voicemail left is not sent to the client. In asterisk 1.6 you need to add a line to the general settings of your voicemail.conf that says 'pollmailboxes=yes' so that asterisk will check the mailboxes for changes that were caused by other programs. It is no by default. Since you are using trixbox you'll probably need to add that line to an include file somewhere. I'm sure if you search the trixbox forums for 'pollmailboxes' you'll find out where it needs to go. -Evan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 MWI issue
I believe that extension 7000 offers dialup access to voicemail. This might be different on your installation. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ilker Aktuna Sent: Tuesday, September 01, 2009 12:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk MWI issue Hi Danny, Thanks for your suggestion. How can I delete voicemail through Asterisk (without using Trixbox GUI which is in fact AMPortal) ? Thanks, ilker - Original Message - From: Danny Nicholas mailto:da...@debsinc.com To: 'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion' Sent: Tuesday, September 01, 2009 8:28 PM Subject: Re: [asterisk-users] Asterisk MWI issue It's (IMO) a TRIXBOX bug. To verify this, leave yourself a voicemail, then delete it through Asterisk. If the behavior is as you expect, then you need to see what gets zapped in Asterisk that TB does not Zap. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] espeak app for asterisk 1.6
I have written a module for asterisk that uses the eSpeak
speech synthesizer (http://espeak.sourceforge.net/) to
render text to speech. The source is available here:
http://zaf.github.com/Asterisk-eSpeak/
It's similar to app_festival and app_flite.
It's only tested against asterisk 1.6.1 on x86 Linux but it must be
working for other 1.6 branches too. Comments, fixes and suggestion are
welcome.
===
Espeak For Asterisk 1.6
===
This provides the Espeak dialplan application, which allows you to use
the Espeak speech synthesizer with Asterisk. This module invokes the Espeak TTS
engine
locally, and uses it to render text to speech.
Requirements
Asterisk 1.6 header files
Espeak libraries and header files
**It is recommended to use espeak version 1.41.01 or newer.
Earlier version of epseak had an file descriptor leak that could
cause asterisk to crash. If upgrading is not an option patch your
current
version of epseak with the espeak.patch provided here.
libsndfile libraries and header files
libresample libraries and header files
Installation
$ make
$ make install
To install the sample configuration file, issue the following command after
the 'make install' command:
$ make samples
-
Usage
-
Espeak(text[,intkeys,language]): This will invoke the eSpeak TTS engine,
send a text string, get back the resulting waveform and play it to
the user, allowing any given interrupt keys to immediately terminate
and return.
Examples
dialplan sample code for your extensions.conf
;Espeak Demo
exten = 1234,1,Answer()
;;Play mesage using default language as set in
espeak.conf
exten = 1234,n,Espeak(This is a simple espeak test in
english.,any,)
;;Play message in Spanish
exten = 1234,n,Espeak(Esta es una simple prueba espeak en
español.,any,es)
;;Play message in Greek
exten = 1234,n,Espeak(ÎÏ
ÏÏ ÎµÎ¯Î½Î±Î¹ Îνα αÏλÏ
ÏÎÏÏ ÏοÏ
espeak ÏÏα ελληνικά.,any,el)
;;Read a text file from disk (relative to the channel
language)
;;and play it with espeak using the asterisk channel
language.
exten = 1234,n,ReadFile(MYTEXT=/path/${LANGUAGE}/myfile,200)
exten = 1234,n,Espeak(${MYTEXY},any,${LANGUAGE})
exten = 1234,n,Hangup()
---
License
---
The Espeak module for asterisk is distributed under the GNU General Public
License v2. See COPYING for details.
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Re: [asterisk-users] SIP and other phones other then local network
On 2/09/09 2:28 AM, Pascal Bruno wrote: For example if it was Alex to reply to that msg, i would feel bad for this guy, because Alex would make him feel like if he cannot do this by himself or use google to find that answer by himself, he does not belong to that list. He would never give him a chance and try to help him. :) That's what I'm here for :) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] MeetMe and dedicated conference room phone
I've googled and not quite found what I need, so... I have a conference room phone that I would like to make behave as follows: - when a call comes to that extension: answer the call put the call in a static MeetMe room with option 'w' ring the phone by SIP and when the phone picks up, put it in the same MeetMe room as the marked call. if subsequent calls come in, they are put in the same room. Is this difficult? Part two, probably harder: - when a call goes out from this SIP phone: save whatever number was dialled put the call in a static MeetMe room (same as above is fine) call the saved number and connect that call to the room. Any ideas? -dsr- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 MWI issue
Ok; when I dial voicemail box and delete the voicemail then I receive the SIP NOTIFY. But TB does not do anything special. Trixbox is kind of Asterisk + AMPortal. If you say, AMPortal does not update Asterisk it is not correct because if I'm on the console of Asterisk, I see the following lines when I delete VM from GUI: == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 Which means, AMPortal is updating Asterisk about the deleted messages. What else should I check ? How can I understand if this is an Asterisk or AMPortal problem ? - Original Message - From: Danny Nicholas To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, September 01, 2009 9:29 PM Subject: Re: [asterisk-users] Asterisk MWI issue I believe that extension 7000 offers dialup access to voicemail. This might be different on your installation. -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ilker Aktuna Sent: Tuesday, September 01, 2009 12:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk MWI issue Hi Danny, Thanks for your suggestion. How can I delete voicemail through Asterisk (without using Trixbox GUI which is in fact AMPortal) ? Thanks, ilker - Original Message - From: Danny Nicholas To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, September 01, 2009 8:28 PM Subject: Re: [asterisk-users] Asterisk MWI issue It's (IMO) a TRIXBOX bug. To verify this, leave yourself a voicemail, then delete it through Asterisk. If the behavior is as you expect, then you need to see what gets zapped in Asterisk that TB does not Zap. -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 MWI issue
On Mon, 31 Aug 2009, ilker Aktuna wrote: When a voicemail is left on the extension, a SIP NOTIFY message is Sent to my SIP client and the MWI is received ok. This is good. But when I delete all Voicemail through AMPortal, SIP NOTIFY message notifying that there is no voicemail left is not sent to the client. In asterisk 1.6 you need to add a line to the general settings of your voicemail.conf that says 'pollmailboxes=yes' so that asterisk will check the mailboxes for changes that were caused by other programs. It is no by default. Since you are using trixbox you'll probably need to add that line to an include file somewhere. I'm sure if you search the trixbox forums for 'pollmailboxes' you'll find out where it needs to go. -Evan I've just added that line to voicemail.conf and reloaded configs. Unfortunately, nothing changed. :( Please keep suggesting something, otherwise I'll go desperate :( ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 MWI issue
Any change to voicemail requires a restart of Asterisk, not just a file reload. Trixbox is doing something; just too much or not enough; -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ilker Aktuna Sent: Tuesday, September 01, 2009 3:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk MWI issue On Mon, 31 Aug 2009, ilker Aktuna wrote: When a voicemail is left on the extension, a SIP NOTIFY message is Sent to my SIP client and the MWI is received ok. This is good. But when I delete all Voicemail through AMPortal, SIP NOTIFY message notifying that there is no voicemail left is not sent to the client. In asterisk 1.6 you need to add a line to the general settings of your voicemail.conf that says 'pollmailboxes=yes' so that asterisk will check the mailboxes for changes that were caused by other programs. It is no by default. Since you are using trixbox you'll probably need to add that line to an include file somewhere. I'm sure if you search the trixbox forums for 'pollmailboxes' you'll find out where it needs to go. -Evan I've just added that line to voicemail.conf and reloaded configs. Unfortunately, nothing changed. :( Please keep suggesting something, otherwise I'll go desperate :( ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 MWI issue
Any change to voicemail requires a restart of Asterisk, not just a file reload. Trixbox is doing something; just too much or not enough; Ok; I've also restarted Asterisk. Doesn't change. Like I said ; Trixbox is not magic. It's just Asterisk plus AMPortal. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Digium PRI cards for data usage?
- robert boardman robert.board...@gmail.com wrote: Do you have to set aside kines for the data channels or can you have dynamic data channels, for example ISDN dialup internet backup? Robb Thats a VERY good question. Many of the circuits I've worked with are dynamically allocated for voice or data depending on voice demands. Will Zaptel/Dahdi accomadate this type of scenario or do the channels have to be statically assigned? Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 MWI issue
On 1 Sep 2009, at 21:49, ilker Aktuna wrote: Any change to voicemail requires a restart of Asterisk, not just a file reload. Trixbox is doing something; just too much or not enough; Ok; I've also restarted Asterisk. Doesn't change. Like I said ; Trixbox is not magic. It's just Asterisk plus AMPortal. Plus a whole load of other stuff. (And amportal has been FreePBX for some time). S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 MWI issue
On 1 Sep 2009, at 21:12, ilker Aktuna wrote: I see the following lines when I delete VM from GUI: == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 Which means, AMPortal is updating Asterisk about the deleted messages. What makes you say that? It could be doing anything. S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Digium PRI cards for data usage?
Tim Nelson wrote: Thats a VERY good question. Many of the circuits I've worked with are dynamically allocated for voice or data depending on voice demands. Will Zaptel/Dahdi accomadate this type of scenario or do the channels have to be statically assigned? Zaptel/DAHDI are not protocol stacks. Doing dynamic channel allocation requires assistance from an application; Asterisk includes an application called DAHDIRAS which can be used to build a RAS-style dial-in server to provide PPP connections (including multilink PPP, I believe). It's not commonly used, but it's been around for a long time and the last time I heard about someone using it did in fact work for them... Now, if you instead of have some sort of non-dial-up method of dynamically changing the number of channels assigned to the HDLC link, I don't know of anything that will support that over DAHDI; I'm not even sure the Linux HDLC network stack can handle that. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] espeak app for asterisk 1.6
Steve Edwards wrote: On Tue, 1 Sep 2009, Lefteris Zafiris wrote: I have written a module for asterisk that uses the eSpeak speech synthesizer (http://espeak.sourceforge.net/) to render text to speech. The source is available here: http://zaf.github.com/Asterisk-eSpeak/ I hope it sounds a whole lot better in practice than it does on their sample available at http://espeak.sourceforge.net/samples/raven.ogg Cepstral's Allison font is miles ahead. Actually it sounds like that demo. You can improve it a bit by altering the voice settings (speed pitch etc) in the config but it cannot match Cepstrals solution. Flite (in which Cepstral is based) gives a better sound but its limited in voice support (i think it supports only English) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 MWI issue
Steve is correct (not a surprise ); all this says is that the manager started and stopped. It gives no clue what if anything the manager actually did. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Tuesday, September 01, 2009 4:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk MWI issue On 1 Sep 2009, at 21:12, ilker Aktuna wrote: I see the following lines when I delete VM from GUI: == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 Which means, AMPortal is updating Asterisk about the deleted messages. What makes you say that? It could be doing anything. S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] espeak app for asterisk 1.6
On Tue, 1 Sep 2009, Lefteris Zafiris wrote: I have written a module for asterisk that uses the eSpeak speech synthesizer (http://espeak.sourceforge.net/) to render text to speech. The source is available here: http://zaf.github.com/Asterisk-eSpeak/ I hope it sounds a whole lot better in practice than it does on their sample available at http://espeak.sourceforge.net/samples/raven.ogg Cepstral's Allison font is miles ahead. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 MWI issue
On 1 Sep 2009, at 21:49, ilker Aktuna wrote: Any change to voicemail requires a restart of Asterisk, not just a file reload. Trixbox is doing something; just too much or not enough; Ok; I've also restarted Asterisk. Doesn't change. Like I said ; Trixbox is not magic. It's just Asterisk plus AMPortal. Plus a whole load of other stuff. (And amportal has been FreePBX for some time). Ok; look, what do you want to prove ? I am not your enemy. I am not a Trixbox developer, I am not someone who is trying to find problems on Asterisk. I am just someone who has a problem and tries to find a solution. Trixbox is not a software itself. It is just a combination of softwares. In my case, there are only two of them involved : Asterisk and AMPortal. Either one of them is the buggy one. And I don't really care what AMPortal was called once upon a time. I see the following lines exactly when I delete a voicemail on the AMPortal GUI. So it points me that AMPortal is updating Asterisk about deleted items. == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 Now, if you wanna help, just tell me what to do. Otherwise, please shut up so that someone who really cares might help me. Sorry for my language but I am really pissed off with some of your answers. (I am sorry everyone for disturbing the mailing list, if you ever feel that way) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Digium PRI cards for data usage?
- Kevin P. Fleming kpflem...@digium.com wrote: Tim Nelson wrote: Thats a VERY good question. Many of the circuits I've worked with are dynamically allocated for voice or data depending on voice demands. Will Zaptel/Dahdi accomadate this type of scenario or do the channels have to be statically assigned? Zaptel/DAHDI are not protocol stacks. Doing dynamic channel allocation requires assistance from an application; Asterisk includes an application called DAHDIRAS which can be used to build a RAS-style dial-in server to provide PPP connections (including multilink PPP, I believe). It's not commonly used, but it's been around for a long time and the last time I heard about someone using it did in fact work for them... Ah yes... I've heard of DAHDIRAS but never looked into it. Thanks. Now, if you instead of have some sort of non-dial-up method of dynamically changing the number of channels assigned to the HDLC link, I don't know of anything that will support that over DAHDI; I'm not even sure the Linux HDLC network stack can handle that. Good point. My primary interest in starting this thread was simply to see if I could use the cards for data access since I have a couple lying around and would like to setup a 1.5mbit T1 circuit between them for data usage. My last post was more of a curiosity type thing. :-) Thanks for the pointers. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] mISDN NT mode config setting
Hi, I am struggling to get plain Cologne chip cards to run in NT mode, runs nice in TE mode despite the error message: login as: root r...@192.168.2.22's password: Last login: Tue Sep 1 23:09:24 2009 from 192.168.2.50 Welcome to Elastix misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX (maybe there is already a PBX running?) Port 2: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX (maybe there is already a PBX running?) Port 3: NT-mode BRI S/T interface port (for phones) - Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX (maybe there is already a PBX running?) Port 4: NT-mode BRI S/T interface port (for phones) - Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX (maybe there is already a PBX running?) I do: etc/misdn-init.conf card=1,hfcpci card=2,hfcpci card=3,hfcpci card=4,hfcpci te_ptmp=1,2 nt_ptp=3,4 --- etc/misdn.conf mISDNconf module poll=128 debug=0 timer=nohfcmulti/module module debug=0 options=0mISDN_dsp/module devnode user=asterisk group=asterisk mode=660mISDN/devnode card type=hfcpci port mode=te link=ptmp1/port /card card type=hfcpci port mode=te link=ptmp1/port /card card type=hfcpci dtmf=yes crystalclock=yes port mode=nt link=ptp1/port /card card type=hfcpci dtmf=yes crystalclock=yes port mode=nt link=ptp1/port /card /mISDNconf -- etc/asterisk/misdn.conf ;Tried with both config's no change misdn_init=/etc/misdn-init.conf ;misdn_init=/etc/mISDN.conf [trunks] ports=1,2 context=from-trunk msns=* [NTports] context=from-internal ports=3,4 msns=* --- I am using Elastix 1.5 not updated Any tips? Where can I find documentation? Thank you! HB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 MWI issue
Here's another piece of the puzzle - check for a file /var/log/asterisk/manevents.log - that should tell you exactly what Trixbox is telling the manager to do. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ilker Aktuna Sent: Tuesday, September 01, 2009 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk MWI issue On 1 Sep 2009, at 21:49, ilker Aktuna wrote: Any change to voicemail requires a restart of Asterisk, not just a file reload. Trixbox is doing something; just too much or not enough; Ok; I've also restarted Asterisk. Doesn't change. Like I said ; Trixbox is not magic. It's just Asterisk plus AMPortal. Plus a whole load of other stuff. (And amportal has been FreePBX for some time). Ok; look, what do you want to prove ? I am not your enemy. I am not a Trixbox developer, I am not someone who is trying to find problems on Asterisk. I am just someone who has a problem and tries to find a solution. Trixbox is not a software itself. It is just a combination of softwares. In my case, there are only two of them involved : Asterisk and AMPortal. Either one of them is the buggy one. And I don't really care what AMPortal was called once upon a time. I see the following lines exactly when I delete a voicemail on the AMPortal GUI. So it points me that AMPortal is updating Asterisk about deleted items. == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 Now, if you wanna help, just tell me what to do. Otherwise, please shut up so that someone who really cares might help me. Sorry for my language but I am really pissed off with some of your answers. (I am sorry everyone for disturbing the mailing list, if you ever feel that way) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 MWI issue
On 1 Sep 2009, at 22:30, ilker Aktuna wrote: Now, if you wanna help, just tell me what to do. Otherwise, please shut up so that someone who really cares might help me. Looking at your history, I think you are making enemies of those who would help. I'll leave you to it. Best of luck with your Trixbox problem. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 MWI issue
On 1 Sep 2009, at 22:37, Danny Nicholas wrote: Here's another piece of the puzzle - check for a file /var/log/asterisk/manevents.log - that should tell you exactly what Trixbox is telling the manager to do. Can debug manager from CLI as well. Should give a few clues. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 MWI issue
I don't have this file under /var/log/asterisk When I delete a VM, only log file named full is updated and the only update is: [Sep 2 01:18:13] VERBOSE[29356] logger.c: == Manager 'admin' logged on from 127.0.0.1 [Sep 2 01:18:14] VERBOSE[29356] logger.c: == Manager 'admin' logged off from 127.0.0.1 [Sep 2 01:18:14] VERBOSE[29358] logger.c: == Manager 'admin' logged on from 127.0.0.1 [Sep 2 01:18:14] VERBOSE[29358] logger.c: == Manager 'admin' logged off from 127.0.0.1 Here's another piece of the puzzle - check for a file /var/log/asterisk/manevents.log - that should tell you exactly what Trixbox is telling the manager to do. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ilker Aktuna Sent: Tuesday, September 01, 2009 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk MWI issue On 1 Sep 2009, at 21:49, ilker Aktuna wrote: Any change to voicemail requires a restart of Asterisk, not just a file reload. Trixbox is doing something; just too much or not enough; Ok; I've also restarted Asterisk. Doesn't change. Like I said ; Trixbox is not magic. It's just Asterisk plus AMPortal. Plus a whole load of other stuff. (And amportal has been FreePBX for some time). Ok; look, what do you want to prove ? I am not your enemy. I am not a Trixbox developer, I am not someone who is trying to find problems on Asterisk. I am just someone who has a problem and tries to find a solution. Trixbox is not a software itself. It is just a combination of softwares. In my case, there are only two of them involved : Asterisk and AMPortal. Either one of them is the buggy one. And I don't really care what AMPortal was called once upon a time. I see the following lines exactly when I delete a voicemail on the AMPortal GUI. So it points me that AMPortal is updating Asterisk about deleted items. == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 Now, if you wanna help, just tell me what to do. Otherwise, please shut up so that someone who really cares might help me. Sorry for my language but I am really pissed off with some of your answers. (I am sorry everyone for disturbing the mailing list, if you ever feel that way) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 2009 - October 13 - 15 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 MWI issue
I'm only top posting to keep the flow going. Otherwise this would get messy. ilker - you should consider bottom posting to not raise the ire of others on the list. This may be a silly question, but do you have mailbox= filled in with the extension's number on the SIP extension page? If not asterisk will not generate the INFO. j On Wed, 2 Sep 2009, ilker Aktuna wrote: I don't have this file under /var/log/asterisk When I delete a VM, only log file named full is updated and the only update is: [Sep 2 01:18:13] VERBOSE[29356] logger.c: == Manager 'admin' logged on from 127.0.0.1 [Sep 2 01:18:14] VERBOSE[29356] logger.c: == Manager 'admin' logged off from 127.0.0.1 [Sep 2 01:18:14] VERBOSE[29358] logger.c: == Manager 'admin' logged on from 127.0.0.1 [Sep 2 01:18:14] VERBOSE[29358] logger.c: == Manager 'admin' logged off from 127.0.0.1 Here's another piece of the puzzle - check for a file /var/log/asterisk/manevents.log - that should tell you exactly what Trixbox is telling the manager to do. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ilker Aktuna Sent: Tuesday, September 01, 2009 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk MWI issue On 1 Sep 2009, at 21:49, ilker Aktuna wrote: Any change to voicemail requires a restart of Asterisk, not just a file reload. Trixbox is doing something; just too much or not enough; Ok; I've also restarted Asterisk. Doesn't change. Like I said ; Trixbox is not magic. It's just Asterisk plus AMPortal. Plus a whole load of other stuff. (And amportal has been FreePBX for some time). Ok; look, what do you want to prove ? I am not your enemy. I am not a Trixbox developer, I am not someone who is trying to find problems on Asterisk. I am just someone who has a problem and tries to find a solution. Trixbox is not a software itself. It is just a combination of softwares. In my case, there are only two of them involved : Asterisk and AMPortal. Either one of them is the buggy one. And I don't really care what AMPortal was called once upon a time. I see the following lines exactly when I delete a voicemail on the AMPortal GUI. So it points me that AMPortal is updating Asterisk about deleted items. == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 Now, if you wanna help, just tell me what to do. Otherwise, please shut up so that someone who really cares might help me. Sorry for my language but I am really pissed off with some of your answers. (I am sorry everyone for disturbing the mailing list, if you ever feel that way) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 2009 - October 13 - 15 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 2009 - October 13 - 15 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] chan_dahdi.so fails to load : Inappropriate ioctl for device
Aloha, I'm not sure why I'm getting this error, but I can't seem to get chan_dahdi to load. SIP IAX2 are working fine. Debian 4 w/ 2.6.28 kernel. Asterisk 1.6.1.5, dahdi-linux 2.2.0.2, dahdi-tools-2.2.0 CLI module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. [Sep 1 10:57:51] WARNING[31696]: pbx.c:4550 ast_register_application2: Already have an application 'DAHDISendKeypadFacility' [Sep 1 10:57:51] ERROR[31696]: chan_dahdi.c:8786 mkintf: Unable to get parameters: Inappropriate ioctl for device [Sep 1 10:57:51] ERROR[31696]: chan_dahdi.c:14170 build_channels: Unable to register channel '1' # dahdi_scan [1] active=yes alarms=OK description=Wildcard AEX410 Board 1 name=WCTDM/0 manufacturer=Digium devicetype=Wildcard AEX410 with VPMADT032 location=PCI Express Bus 11 Slot 09 basechan=1 totchans=4 irq=16 [2] active=yes alarms=UNCONFIGURED description=DAHDI_DUMMY/1 (source: HRtimer) 1 name=DAHDI_DUMMY/1 manufacturer= devicetype=DAHDI Dummy Timing location= basechan=5 totchans=0 irq=0 Note, I have compiled DAHDI 2.2.0.2 but it still shows 2.1.0.4 in the tool. Version bug? If it should say 2.2.0.2, then that could be my problem. But how do I correct that? # dahdi_cfg -vvv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.1.0.4 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Echo Canceler: none) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Echo Canceler: none) (Slaves: 04) 4 channels to configure. Setting echocan for channel 1 to none Setting echocan for channel 2 to none Setting echocan for channel 3 to none Setting echocan for channel 4 to none I have looked up everything I can about this problem, and nothing has lead me to a solution. Any help would be great appreciated. Thanks, Herb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] chan_dahdi.so fails to load : Inappropriate ioctl for device
Forgot to include this info in my first post. # cat /proc/dahdi/* Span 1: WCTDM/0 Wildcard AEX410 Board 1 (MASTER) IRQ misses: 1 1 WCTDM/0/0 FXSKS RED 2 WCTDM/0/1 FXSKS RED 3 WCTDM/0/2 FXOKS 4 WCTDM/0/3 FXOKS I don't have the analog lines attached to the card yet, hence the RED alarms. Span 2: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: HRtimer) 1 Thanks, Herb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Configuring Parallel SIP Trunks
Hi, I'm trying to configure 2 parallel sip trunks between 2 boxes. However I seem to have the problem that when making a call from Box 2 to Box 1, it sometimes says authentication failed because it is using the username of the other trunk. Here's my configuration: Box 1: [dp-dp2] type=peer username=dp-dp2 secret=mysecret qualify=yes host=box.2.ip.address context=from-internal [e911-dp2] context=from-pstn host=box.2.ip.address qualify=yes secret=mysecret2 type=peer username=e911-dp2 Box 2: [dp-dp2] host=box.1.ip.address qualify=yes type=peer username=dp-dp2 secret=mysecret context=from-pstn [e911-dp2] context=from-internal host=box.1.ip.address qualify=yes secret=mysecret2 type=peer username=e911-dp2 If I have both trunks up, I'll see in the log on Box 1, when calling from Box 2 - Box 1: username mismatch, have e911-dp2, digest has dp-dp2 How can I get both to co-exist? Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] DAHDI selective install
Is there any way to not install all DAHDI drivers? All that I need is the dummy driver for timming purposes. Thanks, Valter ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Inquiry:Problem with Call Parking
And now that the whole world of Asterisk has your sip user ids and passwords, you should change all of the passwords that are in that file and yes, change the passwords in all your phones. Lyle Giese LCR Computer Services, Inc. hadi motamedi wrote: Thank you for your reply . Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Regards H.Motamedi On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney) john@compuware.com mailto:john@compuware.com wrote: Just a quick guess - is it because you did not program your Polycom digit plan properly in sip.cfg? From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Tuesday, 1 September 2009 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inquiry:Problem with Call Parking Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my features.conf . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let me know what extra config needs to be done for putting it into operation ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 2009 - October 13 - 15 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 2009 - October 13 - 15 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] Configuring Parallel SIP Trunks
On Tue, 1 Sep 2009, James Lamanna wrote: Hi, I'm trying to configure 2 parallel sip trunks between 2 boxes. However I seem to have the problem that when making a call from Box 2 to Box 1, it sometimes says authentication failed because it is using the username of the other trunk. Here's my configuration: Box 1: [dp-dp2] type=peer username=dp-dp2 secret=mysecret qualify=yes host=box.2.ip.address context=from-internal [e911-dp2] context=from-pstn host=box.2.ip.address qualify=yes secret=mysecret2 type=peer username=e911-dp2 Box 2: [dp-dp2] host=box.1.ip.address qualify=yes type=peer username=dp-dp2 secret=mysecret context=from-pstn [e911-dp2] context=from-internal host=box.1.ip.address qualify=yes secret=mysecret2 type=peer username=e911-dp2 If I have both trunks up, I'll see in the log on Box 1, when calling from Box 2 - Box 1: username mismatch, have e911-dp2, digest has dp-dp2 How can I get both to co-exist? Thanks. -- James Hi James, Try changing the host idents to 'dynamic'. I think the IP match may come first, though I don't know why it would be intermittent and not always a problem. If that works you could limit each entry with appropriate permit/deny statements. Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] CDR to Postgres Centos
Hello Barry I am using asterisk 1.6.1.2 ... Yeah the database was created at that time . because if the database is not created then it gives another error i remeber like .. database not found or not accepting connections... The problem is solved automatically ... I have done nothing (restart etc not included in nothing ;)) It is working fine Thanks On Tue, Sep 1, 2009 at 5:26 PM, Barry L. Kline blkl...@attglobal.netwrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 ABBAS SHAKEEL wrote: Can i know which querry is executed to insert record to database... i am asking this because of [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:309 pgsql_log: Failed to insert call detail record into database! [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:310 pgsql_log: Reason: ERROR: syntax error at or near ) at character 17 [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:311 pgsql_log: Connection may have been lost... attempting to reconnect. [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:314 pgsql_log: Connection reestablished. [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:320 pgsql_log: HARD ERROR! Attempted reconnection failed. DROPPING CALL RECORD! Which version of Asterisk are you using? Did you create the PG database for Asterisk? Have you confirmed that you can connect to it using the CLI psql with the appropriate credentials? There are a few steps ahead of where you are before we worry about this particular problem. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKnRL4CFu3bIiwtTARAnwNAJ9+CiWdtq17DRSqelNl7bsN5pS32gCeIn+l VNyWYBauMOBvVMhyGUeP/Pk= =G9NP -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] AMI Originate Commands executed in sequential Order problem
Hi, I noticed that asterisk manager interface will only accept the originate commands in sequential order. For example, if I want to ring two extensions through the AMI, and while first extension is ringing, AMI won't execute and ring second extension until first extension has answered the call. Anybody has any ideas as I had the same results even tested with telnet commands to AMI interface. Thanks Best Regards, Johnny Xing -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-users-requ...@lists.digium.com Sent: 2009年9月1日 15:49 To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 62, Issue 1 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Inquiry:Problem with Call Parking (hadi motamedi) 2. Re: Asterisk Web Meetme module not loading (Glen) 3. Re: Inquiry:Problem with Call Parking (Lee, John (Sydney)) 4. Re: Asterisk Web Meetme module not loading (Matt Riddell) 5. Re: Inquiry:Problem with Call Parking (hadi motamedi) 6. Re: Inquiry:Problem with Call Parking (Matt Riddell) 7. Re: Inquiry:Problem with Call Parking (Darrick Hartman) 8. Re: Asterisk Web Meetme module not loading (Glen Ganderton) 9. Re: Inquiry:Problem with Call Parking (Lee, John (Sydney)) 10. Re: Asterisk Web Meetme module not loading (Matt Riddell) 11. Re: Inquiry:Problem with Call Parking (Paul Hales) 12. Re: Asterisk Web Meetme module not loading (Matt Riddell) 13. Re: Asterisk Web Meetme module not loading (Matt Riddell) 14. Re: Asterisk Web Meetme module not loading (Matt Riddell) 15. Re: Asterisk Web Meetme module not loading (Glen) 16. Re: Asterisk Web Meetme module not loading (Matt Riddell) 17. jitterbuffer for chan_sip on asterisk 1.2 (James Mutuku) 18. Inquiry:Problem with VoiceMail (hadi motamedi) 19. Re: CDR to Postgres Centos (ABBAS SHAKEEL) 20. SIP and other phones other then local network (ABBAS SHAKEEL) -- Message: 1 Date: Tue, 1 Sep 2009 05:39:24 +0100 From: hadi motamedi motamed...@gmail.com Subject: [asterisk-users] Inquiry:Problem with Call Parking To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 74aa57df0908312139v57fd26f5v81961f10873e3...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my features.conf . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let me know what extra config needs to be done for putting it into operation ? Regards H.Motamedi -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090901/3cc8f5 02/attachment-0001.htm -- next part -- A non-text attachment was scrubbed... Name: features.conf Type: application/octet-stream Size: 4820 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090901/3cc8f5 02/attachment-0001.obj -- Message: 2 Date: Tue, 01 Sep 2009 14:54:43 +1000 From: Glen glengander...@gmail.com Subject: Re: [asterisk-users] Asterisk Web Meetme module not loading To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 4a9ca913.1070...@gmail.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed Matt Riddell wrote: On 1/09/09 4:31 PM, Glen wrote: Matt Riddell wrote: On 31/08/09 2:33 PM, Glen wrote: I have asterisk 1.4.21 and web meetme (latest release 3.1) I have also installed the latest versions of mysql and php. I followed the readme file that came with the web meetme app and everything seemed to go fine up until I realised the module wasnt being loaded. When I stop asterisk and try to start it, it errors out and does not load and I get the following message: Parsing '/etc/asterisk/cbmysql.conf': Found asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_cbmysql.so: undefined symbol: mysql_init Likely you don't have mysql-devel libraries installed - though I wonder how
Re: [asterisk-users] AMI Originate Commands executed in sequential Order problem
On 2/09/09 3:22 PM, johnny_xing wrote: Hi, I noticed that asterisk manager interface will only accept the originate commands in sequential order. For example, if I want to ring two extensions through the AMI, and while first extension is ringing, AMI won't execute and ring second extension until first extension has answered the call. Anybody has any ideas as I had the same results even tested with telnet commands to AMI interface. Wow, funky font! Use asyncronous mode -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Configuring Parallel SIP Trunks
2 sep 2009 kl. 02.44 skrev James Lamanna: Hi, I'm trying to configure 2 parallel sip trunks between 2 boxes. However I seem to have the problem that when making a call from Box 2 to Box 1, it sometimes says authentication failed because it is using the username of the other trunk. Here's my configuration: Box 1: [dp-dp2] type=peer username=dp-dp2 secret=mysecret qualify=yes host=box.2.ip.address context=from-internal [e911-dp2] context=from-pstn host=box.2.ip.address qualify=yes secret=mysecret2 type=peer username=e911-dp2 Box 2: [dp-dp2] host=box.1.ip.address qualify=yes type=peer username=dp-dp2 secret=mysecret context=from-pstn [e911-dp2] context=from-internal host=box.1.ip.address qualify=yes secret=mysecret2 type=peer username=e911-dp2 If I have both trunks up, I'll see in the log on Box 1, when calling from Box 2 - Box 1: username mismatch, have e911-dp2, digest has dp-dp2 How can I get both to co-exist? Well, you have to learn how Asterisk matches incoming calls for peers. The peer matching is done on ip/port and will match the first one in the internal list, which is the last one in the configuration. All incoming calls will come to e911-dp2 in your configuration, since both peers has the same ip address and port. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
