Re: [asterisk-users] SIP DNS SRV
On 11/08/2010 09:50 PM, Jonas Kellens wrote: Hello, SIP DNS SRV records are not working. My Grandstream uses the SRV records to find the first Asterisk server to register to. This works. But when I shut down the Asterisk proces on server 1 and I restart my GXP 2010, the phone does not register to server 2... No mather how long I wait, there is no registration coming in... When I start the Asterisk proces again on server 1, then here registration comes in. Kind regards, Jonas. More info : [jo...@jonas ~]$ host -t srv _SIP._udp.sip10.domain.tld _SIP._udp.sip10.domain.tld has SRV record 25 10 5060 sip2.domain.tld. _SIP._udp.sip10.domain.tld has SRV record 5 10 5060 sip1.domain.tld. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
On Mon, 8 Nov 2010, Bruce B wrote: Yes, it is a small office. I am familiar with pfSense. I am not sure if firewall on Astlinux is as versatile and flexible. But also, I am wondering if with all those attacks around now-a-days if the box will be able to handle 5 extensions, voicemail, IVR, firewall, DHCP, openvpn all together. I've benchmarked an Alix board with a 500MHz processor to 80 concurrent calls handling media. They're the mainstay of my small office VoIp only PBX range right now where I limit them to 60 extensions. (the real limitation on number of calls is their broadband bandwidth). Storing voicemail and call recording won't be an issue for you - but do get a fast CF card. http://unicorn.drogon.net/cutie.jpg What you need to do is learn Linux networking and iptables - then you won't need pfsense, etc. Install a good text-only distribution and you're done. e.g. Debian Lenny in text-only mode. To get he best from the hardware then you'll need a custom kernel, but that's no big deal. However the thing that will kill it is multiple VPN terminations - unless you can persuade the system to use the on-board AES crypto engine, but I regularly use ssh into my systems without any detriment, so you could use OpenVPN, etc. I am considering making my boxes into a router and handle PPPoE too, then they can do proper traffic shaping, etc. They're more than capable. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Voicemail Realtime and 'VirtualBoxing'
Hello
I'm about to set up a voicemail system for multiple wholesale customers.
So I use a realtime mysql config for the mailboxes.
All single mailboxes have their information about the number, emailaddress,
password in the database. This works fine.
Now the notification emails of course should be customized per wholesale
customer.
I added a 'mandate' table to the database and get this field by an AGI script
before calling VoiceMail to get the correct language and context name for
this particular mailbox. Let's call them company1 and company2
Then I do:
exten = s,n,AGI(getmandate.agi)
exten = s,n,Set(CHANNEL(language)=${MANDATELANG})
exten = s,n,VoiceMail(0${CALLERID(rdnis):2...@${mandate},u)
in voicemail.conf I have:
[company1]
serveremail=voicem...@company1.example.com
tz=european
emailsubject=[Customer 1 VM]: Neue Nachricht nummer ${VM_MSGNUM} von
${VM_CIDNUM} in mailbox ${VM_MAILBOX}.
emailbody=some more blabla
[company2]
serveremail=com...@company2.example.com
tz=european
emailsubject=[COMBOX]: New Message from ${VM_CIDNUM} in mailbox ${VM_MAILBOX}.
emailbody=some other blabla
Unfortunately the email settings per voicemail context are ignored, those from
the [general] section are being used.
As far as I found out, I could also put emailsubject and emailbody in the
database, but this would massively increase the size of the database and as
this is not information which will often change this does not need to be
realtime.
Is there a way to have the email settings per voicemail context together with
a realtime vm config?
Mit freundlichen Grüssen
Benoit Panizzon
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Re: [asterisk-users] Festival
Hi, wich version of Asterisk? If is 1.6.2.13, there is a open issue becouse not work https://issues.asterisk.org/view.php?id=17995 R.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP DNS SRV
On 11/09/2010 02:12 PM, Gareth Blades wrote: Jonas Kellens wrote: On 11/08/2010 09:50 PM, Jonas Kellens wrote: Hello, SIP DNS SRV records are not working. My Grandstream uses the SRV records to find the first Asterisk server to register to. This works. But when I shut down the Asterisk proces on server 1 and I restart my GXP 2010, the phone does not register to server 2... No mather how long I wait, there is no registration coming in... When I start the Asterisk proces again on server 1, then here registration comes in. Kind regards, Jonas. More info : [jo...@jonas ~]$ host -t srv _SIP._udp.sip10.domain.tld _SIP._udp.sip10.domain.tld has SRV record 25 10 5060 sip2.domain.tld. _SIP._udp.sip10.domain.tld has SRV record 5 10 5060 sip1.domain.tld. It sounds like the grandstream phones are not fully compliant with the SRV standard. They are probably just looking for the lowest priority entry and hardcoding that to be used all the time internally. If you restart the phone does it work? It might try the 25 priority entry if it cannot initially contact the primary server. The way I test it : - Grandstream turned off. - Stop asterisk server1 (/sbin/service asterisk stop) - Turn on Grandstream (power up) Conclusion : Grandstream does not register. No register coming in on server2. Finally : - Start Asterisk again on server1 (/sbin/service asterisk start) Conclusion : Grandstream registers to server1. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Festival
Dear Asterisk-Users, I installed festival and while trying to connect it to asterisk it comes up with: serverMon Nov 8 18:38:51 2010 : Festival server started on port 1314client(1) Mon Nov 8 18:38:51 2010 : accepted from localhost.localdomainclient(1) Mon Nov 8 18:38:51 2010 : disconnected I tried editing the festival.scm file and came out worse with socket: bind failed Please can anyone help out. Thanks in advance. ABEJIDE, Ayodele A. (CCNA) +2348039269311 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor
So it seems not related to MixMonitor. Are you 100% sure that your PHP-AGi script is not looping somewhere? You should try to understand which is the process that is taken you CPU. On Tue, Nov 9, 2010 at 2:32 PM, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Hi, After disabling MixMonitor, I realize that my CPU saturates as always! What my script PHP-AGI is fairly simple! - I answer a call - Some menus - I send the call to another line $this-exec_dial (SIP/provider/NUMBER, ...) And I was 75-80% using an e4...@2.40ghz! It is not logic ! Please help ! 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com Hi, marked - noticed. I do not know where it comes from, my CPU goes from 2% to 60-70% at a command Dial (sip) + MixMonitor. I have an Intel (R) Core (TM) 2 Duo CPU e4...@2.40ghz 2010/11/5 Norbert Zawodsky norb...@zawodsky.at Am 05.11.2010 10:16, schrieb Mickael MONSIEUR: none ? 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com mailto:mickael.monsi...@gmail.com Hi, Have you noticed a marked increase in CPU load when using MixMonitor? I use PHPAgi and Asterisk 1.6.2.9-2. Mickael. Obviously, if the box has more to do, CPU load will increase. What do you mean with marked ?? Norbet -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Se l'è vera che te me voeuret ben cara Ninin biribimpinpin vegn giò a derví el portell famm pú penà, parabappappà se ti te gh'hee l'amor del tò Marcell che l'è inscí bell vegn giò a derví el portell famm pú penà, parabappappà! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP DNS SRV
Jonas Kellens wrote: On 11/08/2010 09:50 PM, Jonas Kellens wrote: Hello, SIP DNS SRV records are not working. My Grandstream uses the SRV records to find the first Asterisk server to register to. This works. But when I shut down the Asterisk proces on server 1 and I restart my GXP 2010, the phone does not register to server 2... No mather how long I wait, there is no registration coming in... When I start the Asterisk proces again on server 1, then here registration comes in. Kind regards, Jonas. More info : [jo...@jonas ~]$ host -t srv _SIP._udp.sip10.domain.tld _SIP._udp.sip10.domain.tld has SRV record 25 10 5060 sip2.domain.tld. _SIP._udp.sip10.domain.tld has SRV record 5 10 5060 sip1.domain.tld. It sounds like the grandstream phones are not fully compliant with the SRV standard. They are probably just looking for the lowest priority entry and hardcoding that to be used all the time internally. If you restart the phone does it work? It might try the 25 priority entry if it cannot initially contact the primary server. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Thanks for input. Great info. Good to know all this about the router. I see you use a 256MB CF card there. Do you use a USB key stick for storage? Thanks, Bruce On Tue, Nov 9, 2010 at 4:09 AM, Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net wrote: On Mon, 8 Nov 2010, Bruce B wrote: Yes, it is a small office. I am familiar with pfSense. I am not sure if firewall on Astlinux is as versatile and flexible. But also, I am wondering if with all those attacks around now-a-days if the box will be able to handle 5 extensions, voicemail, IVR, firewall, DHCP, openvpn all together. I've benchmarked an Alix board with a 500MHz processor to 80 concurrent calls handling media. They're the mainstay of my small office VoIp only PBX range right now where I limit them to 60 extensions. (the real limitation on number of calls is their broadband bandwidth). Storing voicemail and call recording won't be an issue for you - but do get a fast CF card. http://unicorn.drogon.net/cutie.jpg What you need to do is learn Linux networking and iptables - then you won't need pfsense, etc. Install a good text-only distribution and you're done. e.g. Debian Lenny in text-only mode. To get he best from the hardware then you'll need a custom kernel, but that's no big deal. However the thing that will kill it is multiple VPN terminations - unless you can persuade the system to use the on-board AES crypto engine, but I regularly use ssh into my systems without any detriment, so you could use OpenVPN, etc. I am considering making my boxes into a router and handle PPPoE too, then they can do proper traffic shaping, etc. They're more than capable. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Festival
It is 1.6.2.13 ABEJIDE, Ayodele A. (CCNA) +2348039269311 From: asannu...@gmail.com To: asterisk-users@lists.digium.com Date: Tue, 9 Nov 2010 07:38:44 -0500 Subject: Re: [asterisk-users] Festival Hi, wich version of Asterisk? If is 1.6.2.13, there is a open issue becouse not work https://issues.asterisk.org/view.php?id=17995 R. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor
Hi, After disabling MixMonitor, I realize that my CPU saturates as always! What my script PHP-AGI is fairly simple! - I answer a call - Some menus - I send the call to another line $this-exec_dial (SIP/provider/NUMBER, ...) And I was 75-80% using an e4...@2.40ghz! It is not logic ! Please help ! 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com Hi, marked - noticed. I do not know where it comes from, my CPU goes from 2% to 60-70% at a command Dial (sip) + MixMonitor. I have an Intel (R) Core (TM) 2 Duo CPU e4...@2.40ghz 2010/11/5 Norbert Zawodsky norb...@zawodsky.at Am 05.11.2010 10:16, schrieb Mickael MONSIEUR: none ? 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com mailto:mickael.monsi...@gmail.com Hi, Have you noticed a marked increase in CPU load when using MixMonitor? I use PHPAgi and Asterisk 1.6.2.9-2. Mickael. Obviously, if the box has more to do, CPU load will increase. What do you mean with marked ?? Norbet -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP DNS SRV
Jonas Kellens wrote: On 11/09/2010 02:12 PM, Gareth Blades wrote: Jonas Kellens wrote: On 11/08/2010 09:50 PM, Jonas Kellens wrote: Hello, SIP DNS SRV records are not working. My Grandstream uses the SRV records to find the first Asterisk server to register to. This works. But when I shut down the Asterisk proces on server 1 and I restart my GXP 2010, the phone does not register to server 2... No mather how long I wait, there is no registration coming in... When I start the Asterisk proces again on server 1, then here registration comes in. Kind regards, Jonas. More info : [jo...@jonas ~]$ host -t srv _SIP._udp.sip10.domain.tld _SIP._udp.sip10.domain.tld has SRV record 25 10 5060 sip2.domain.tld. _SIP._udp.sip10.domain.tld has SRV record 5 10 5060 sip1.domain.tld. It sounds like the grandstream phones are not fully compliant with the SRV standard. They are probably just looking for the lowest priority entry and hardcoding that to be used all the time internally. If you restart the phone does it work? It might try the 25 priority entry if it cannot initially contact the primary server. The way I test it : - Grandstream turned off. - Stop asterisk server1 (/sbin/service asterisk stop) - Turn on Grandstream (power up) Conclusion : Grandstream does not register. No register coming in on server2. Finally : - Start Asterisk again on server1 (/sbin/service asterisk start) Conclusion : Grandstream registers to server1. Jonas. Then it looks like the Grandstream phone dont fully support DNS SRV. maybe a firmware update will fix it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SMS Gateway
Hi list, Anyone has some guidance in how can I project a SMS gateway with Asterisk. I mean, some good web link,pdf or something like that? Thanks in advanced!!Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormirandaru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Store CDR (call detail record) to Oracle database
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phuong Hoang Sent: Monday, November 08, 2010 8:31 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Store CDR (call detail record) to Oracle database Hi all, Now i want to store cdr (call detail record) to Oracle database but i don't know how to do .Can anyone help me ? Thanks and best regards. ODBC is going to be your best bet. At some point you might be able to use the MYSQL stuff to talk from Asterisk to Oracle, but for now I'm pretty sure it is only accessible through ODBC. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] scratchy sound on TE410P
On Mon, Nov 08, 2010 at 02:44:26PM -0500, Jeff LaCoursiere wrote: It could be the echo canceller, I had this kind of problem with OSLEC. I also thought the PRI provider was sending clipped audio. I switched to the VPM450 daughterboard and since audio has been crystal clear. What is your setup for echo cancelling? I inherited this board, and don't think it has the echo canceller daughterboard. Is there a way to query for it without taking the machine down? It is loading MG2 otherwise. My problem seemed to be OSLEC specific (Debian stable with zaptel), switching to MG2 made my problems dissapear but overall voice quality is lower IMHO. You could try disabling ec all together and check if clipping still occurs. But it does sound like an operator problem if you get errors. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
On Tue, 9 Nov 2010, Bruce B wrote: Thanks for input. Great info. Good to know all this about the router. I see you use a 256MB CF card there. Do you use a USB key stick for storage? No. Things that stick out of boxes in small offices get broken off. (ie. the type of places that do not have a server room) Everything is stored on the 256MB CF card - which is overkill, I only need 64MB for the OS, etc., but it provides plenty of space for voicemail and additional sound prompts. I provide a 2GB card for people who want to store call recordings. I actually have my own cut-down version of Linux and a full-custom compiled kernel that I use in these things, but you'll get a standard Debian in there if you go for a text-only install, but a 2GB CF cards isn't a big hassle. I don't think it will boot off USB though (but I've never tried) - there is a 44-pin header behind the CF socket, so maybe you can plug a suitable CD-ROM drive in that. When building, I boot them via PXE then use my booted image to write a copy of itself to the flash. The bios is very primitive and of-course there's no video hardware on-board. They run at 5 watts which is nice too. Gordon Thanks, Bruce On Tue, Nov 9, 2010 at 4:09 AM, Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net wrote: On Mon, 8 Nov 2010, Bruce B wrote: Yes, it is a small office. I am familiar with pfSense. I am not sure if firewall on Astlinux is as versatile and flexible. But also, I am wondering if with all those attacks around now-a-days if the box will be able to handle 5 extensions, voicemail, IVR, firewall, DHCP, openvpn all together. I've benchmarked an Alix board with a 500MHz processor to 80 concurrent calls handling media. They're the mainstay of my small office VoIp only PBX range right now where I limit them to 60 extensions. (the real limitation on number of calls is their broadband bandwidth). Storing voicemail and call recording won't be an issue for you - but do get a fast CF card. http://unicorn.drogon.net/cutie.jpg What you need to do is learn Linux networking and iptables - then you won't need pfsense, etc. Install a good text-only distribution and you're done. e.g. Debian Lenny in text-only mode. To get he best from the hardware then you'll need a custom kernel, but that's no big deal. However the thing that will kill it is multiple VPN terminations - unless you can persuade the system to use the on-board AES crypto engine, but I regularly use ssh into my systems without any detriment, so you could use OpenVPN, etc. I am considering making my boxes into a router and handle PPPoE too, then they can do proper traffic shaping, etc. They're more than capable. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor
You think of a loop? This is possible because I use AGISIGHUP=no .. exten = s,1,set(AGISIGHUP=no); exten = s,2,AGI(myapp.agi) ; I will put lines and debug log file ... I do not think that Asterisk archive errors AGI script? 2010/11/9 Marino Punturieri map...@gmail.com So it seems not related to MixMonitor. Are you 100% sure that your PHP-AGi script is not looping somewhere? You should try to understand which is the process that is taken you CPU. On Tue, Nov 9, 2010 at 2:32 PM, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Hi, After disabling MixMonitor, I realize that my CPU saturates as always! What my script PHP-AGI is fairly simple! - I answer a call - Some menus - I send the call to another line $this-exec_dial (SIP/provider/NUMBER, ...) And I was 75-80% using an e4...@2.40ghz! It is not logic ! Please help ! 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com Hi, marked - noticed. I do not know where it comes from, my CPU goes from 2% to 60-70% at a command Dial (sip) + MixMonitor. I have an Intel (R) Core (TM) 2 Duo CPU e4...@2.40ghz 2010/11/5 Norbert Zawodsky norb...@zawodsky.at Am 05.11.2010 10:16, schrieb Mickael MONSIEUR: none ? 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com mailto:mickael.monsi...@gmail.com Hi, Have you noticed a marked increase in CPU load when using MixMonitor? I use PHPAgi and Asterisk 1.6.2.9-2. Mickael. Obviously, if the box has more to do, CPU load will increase. What do you mean with marked ?? Norbet -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Se l'è vera che te me voeuret ben cara Ninin biribimpinpin vegn giò a derví el portell famm pú penà, parabappappà se ti te gh'hee l'amor del tò Marcell che l'è inscí bell vegn giò a derví el portell famm pú penà, parabappappà! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 and Zimbra
Has anyone managed to successfully connect Asterisk to Zimbra using the Jabber service ? I have opened http://issues.asterisk.org/view.php?id=18198 as it keeps failing for me. Am wondering whether it is due to using a self signed cert. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 and Zimbra
--[ UxBoD ]-- wrote: Has anyone managed to successfully connect Asterisk to Zimbra using the Jabber service I did a couple months ago, using GaJim, but haven't been able to reproduce it. I've since moved on to OpenFire for my Jabber server I will be revisiting this again, hopefully before the end of the year, since our company would like our Zimbra NE server to handle it. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS Gateway
Try kannel http://www.kannel.org It' a very good and powerful WAP and SMS gateway. Adolphe Cher-aime From my Iphone On Nov 9, 2010, at 10:35 AM, Flavio Miranda flaviormira...@hotmail.com wrote: Hi list, Anyone has some guidance in how can I project a SMS gateway with Asterisk. I mean, some good web link,pdf or something like that? Thanks in advanced!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormirandaru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Addons for Asterisk 1.8?
On Tue, Nov 9, 2010 at 2:09 AM, Tilghman Lesher tles...@digium.com wrote: On Monday 08 November 2010 16:05:28 Carlos Chavez wrote: On Mon, 2010-11-08 at 16:53 -0500, bakko wrote: The addons are in the same package. Regards - Original Message - From: Carlos Chavez cur...@telecomabmex.com To: Asterisk asterisk-users@lists.digium.com Sent: Monday, November 08, 2010 4:43 PM Subject: [asterisk-users] Addons for Asterisk 1.8? Yes, I spoke before opening the UPGRADE.txt file and reading that they are now included in the same archive. I do not quite understand why they changed the distribution method. I think it is better to have a separate package that you do not have to download every time you upgrade Asterisk. The reason for including it is because when we occasionally need to change the internal API, it's difficult to remember which version of the addons package goes with each version of Asterisk. Additionally, changes to the addons packages rarely got released on time in the past. It makes it a lot easier just to include them directly in the main distribution, default disabled. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Not to mention, they're not _that_ much of an increase in download size ;-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 and Username in Dial
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
Thanks
Olivier
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Re: [asterisk-users] MixMonitor
Not sure, but you can try to increase debug log level and check whether you'll have more details On Tue, Nov 9, 2010 at 4:55 PM, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: You think of a loop? This is possible because I use AGISIGHUP=no .. exten = s,1,set(AGISIGHUP=no); exten = s,2,AGI(myapp.agi) ; I will put lines and debug log file ... I do not think that Asterisk archive errors AGI script? 2010/11/9 Marino Punturieri map...@gmail.com So it seems not related to MixMonitor. Are you 100% sure that your PHP-AGi script is not looping somewhere? You should try to understand which is the process that is taken you CPU. On Tue, Nov 9, 2010 at 2:32 PM, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Hi, After disabling MixMonitor, I realize that my CPU saturates as always! What my script PHP-AGI is fairly simple! - I answer a call - Some menus - I send the call to another line $this-exec_dial (SIP/provider/NUMBER, ...) And I was 75-80% using an e4...@2.40ghz! It is not logic ! Please help ! 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com Hi, marked - noticed. I do not know where it comes from, my CPU goes from 2% to 60-70% at a command Dial (sip) + MixMonitor. I have an Intel (R) Core (TM) 2 Duo CPU e4...@2.40ghz 2010/11/5 Norbert Zawodsky norb...@zawodsky.at Am 05.11.2010 10:16, schrieb Mickael MONSIEUR: none ? 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com mailto:mickael.monsi...@gmail.com Hi, Have you noticed a marked increase in CPU load when using MixMonitor? I use PHPAgi and Asterisk 1.6.2.9-2. Mickael. Obviously, if the box has more to do, CPU load will increase. What do you mean with marked ?? Norbet -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Se l'è vera che te me voeuret ben cara Ninin biribimpinpin vegn giò a derví el portell famm pú penà, parabappappà se ti te gh'hee l'amor del tò Marcell che l'è inscí bell vegn giò a derví el portell famm pú penà, parabappappà! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Se l'è vera che te me voeuret ben cara Ninin biribimpinpin vegn giò a derví el portell famm pú penà, parabappappà se ti te gh'hee l'amor del tò Marcell che l'è inscí bell vegn giò a derví el portell famm pú penà, parabappappà! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2
Is there a way running Trixbox Pro and Aastra 6731i phones to display the name of the extension you are trying to dial? For example, I want to dial John Smith at x4000, I pick up my phone, dial x4000 and it displays John Smith? Thanks --Dovey -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel debugging
Hi, How to enable zaptel debugging? I need to see reverse polarity messages. Thank you, Imran -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
Nobody has any idea why the Caller ID is being overwritten when using Asterisk
Realtime for the SIP users?
Brett Woollum
br...@woollum.com
- Original Message -
From: Brett Woollum br...@woollum.com
To: asterisk-users@lists.digium.com
Sent: Sunday, November 7, 2010 3:08:50 PM GMT -08:00 US/Canada Pacific
Subject: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num)
Problem
Hello,
I am running Asterisk 1.6 with Realtime enabled for my SIP users and peers. The
backend is a MySQL database running through the ODBC backend in Asterisk. At
this point everything works in terms of phones registering, placing calls
between them, etc. However, I am having a problem setting the Caller ID number
whenever I am using the Realtime database for the SIP users/peers. If I use a
static sip.conf configuration instead of the database, everything works fine.
Unfortunately a static sip.conf file won't work in my application.
In this example:
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)}) ;;;PS: This shows the
correct number of 2 on the CLI console...
exten = 412,n,Dial(SIP/412)
Whenever another phone calls extension 412, the call is forwarded to SIP/412
and should have TEST as the CallerID name and 2 as the CallerID number.
But, whenever I am using the realtime backend, the caller ID number always
displays on the destination phone as that phone's username. Meaning, if phone
SIP/412 receives the call from the example above, the caller ID name displayed
is TEST but the caller ID number is always 412.
What could be causing this?
Brett Woollum
br...@woollum.com
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Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
On Tue, Nov 9, 2010 at 6:35 PM, Brett Woollum br...@woollum.com wrote: Nobody has any idea why the Caller ID is being overwritten when using Asterisk Realtime for the SIP users? No, perhaps you can _show_ us the problem. https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Set(SIP/413-0005, CALLERID(num)=2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:2]
NoOp(SIP/413-0005, CallerID(num) is: 2 ) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:3]
Dial(SIP/413-0005, SIP/412) in new stack
[Nov 9 17:33:39] VERBOSE[4175] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] app_dial.c: -- Called 412
[Nov 9 17:33:40] VERBOSE[4175] app_dial.c: -- SIP/412-0006 is ringing
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, 412, 3)
exited non-zero on 'SIP/413-0005'
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Hangup(SIP/413-0005, ) in new stack
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, h, 1)
exited non-zero on 'SIP/413-0005'
As you can see on line 3, CallerID(num) was successfully set to 2.
SIP/412 is dialed next. It receives the call, but with 412 as the Caller ID
number - even though the real source of the call was extension 413. The name I
set in CallerID(name) works fine.
My Extensions.conf for that context:
[sipphones]
exten = 412,1,Set(CALLERID(num)=2)
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)})
exten = 412,n,Dial(SIP/412)
exten = 412,n,NoOp(${CALLERID(num)})
If I disable sippusers and sippeers in extconfig.conf and put 412 and 413 into
sip.conf directly, this code works (ie: the CallerID(num) I set makes it out to
the destination phone properly).
Brett Woollum
br...@woollum.com
- Original Message -
From: Paul Belanger paul.belan...@polybeacon.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 9, 2010 5:18:36 PM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten
CallerID(num) Problem
On Tue, Nov 9, 2010 at 6:35 PM, Brett Woollum br...@woollum.com wrote:
Nobody has any idea why the Caller ID is being overwritten when using
Asterisk Realtime for the SIP users?
No, perhaps you can _show_ us the problem.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
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Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) |
Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger
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[asterisk-users] Asterisk ConfBridge application – Delay in voice path
Hi All, I am running asterisk on Linux machine and trying to use confbridge application. Please have a look at Conf files. sip.conf == [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow = all allow=ulaw allow=alaw defaultexpiry=100 [5001] type=friend nat=yes host=dynamic canreinvite=no context= conferences disallow = all allow=ulaw allow=alaw [5002] type=friend nat=yes host=dynamic canreinvite=no context= conferences disallow = all allow=ulaw allow=alaw [5003] type=friend nat=yes host=dynamic canreinvite=no context= conferences disallow = all allow=ulaw allow=alaw [5004] type=friend nat=yes host=dynamic canreinvite=no context= conferences disallow = all allow=ulaw allow=alaw extensions.conf [general] static = yes writeprotect = no clearglobalvars = no autofallthrough = yes [conferences] exten = 999,1,Answer() exten = 999,n,ConfBridge(conference,M) I have added 4 users to confbridge by dialing 999.From first 3 users [X-lite] no delay in voice path. But 4th user [X-lite] has delay in voice path and it taking nearly 30 Seconds [ some times 1minute also] to reach voice to remaining 3-persons. Once in a while it happening after adding 3rd user itself. Any Idea - Please suggest me. Thanks and Regards, Garge. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
10 nov 2010 kl. 02.38 skrev Brett Woollum:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Set(SIP/413-0005, CALLERID(num)=2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:2]
NoOp(SIP/413-0005, CallerID(num) is: 2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:3]
Dial(SIP/413-0005, SIP/412) in new stack
[Nov 9 17:33:39] VERBOSE[4175] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] app_dial.c: -- Called 412
[Nov 9 17:33:40] VERBOSE[4175] app_dial.c: -- SIP/412-0006 is ringing
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, 412,
3) exited non-zero on 'SIP/413-0005'
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Hangup(SIP/413-0005, ) in new stack
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, h, 1)
exited non-zero on 'SIP/413-0005'
As you can see on line 3, CallerID(num) was successfully set to 2.
SIP/412 is dialed next. It receives the call, but with 412 as the Caller ID
number - even though the real source of the call was extension 413. The name
I set in CallerID(name) works fine.
My Extensions.conf for that context:
[sipphones]
exten = 412,1,Set(CALLERID(num)=2)
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)})
exten = 412,n,Dial(SIP/412)
exten = 412,n,NoOp(${CALLERID(num)})
If I disable sippusers and sippeers in extconfig.conf and put 412 and 413
into sip.conf directly, this code works (ie: the CallerID(num) I set makes it
out to the destination phone properly).
Have you set the fromuser= field in the realtime database?
/O
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Re: [asterisk-users] OT: certificate for softphone
6 nov 2010 kl. 15.30 skrev Hans Witvliet: Hi all, As stated in the subject, slightly off-topic, as it is not directly a Asterisk issue, but more SIP in general Because security in general, and specifically identification becomes more and more a subject for more concern, and Asterisk is capable of doing sip/TLS, i was wondering what more could be done to improve security. Specially softphones, might it be possible to employ etokens or smartcards for holding the certificates needed by TLS? Done before? In the SIP protocol there is support for TLS client certificates, much like in HTTP. Asterisk doesn't support it. You need to put a SIP proxy like Kamailio in front of Asterisk to get this kind of strong authentication. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exceptionally long queue length queuing . . . .
31 okt 2010 kl. 13.43 skrev Paul Belanger: On Sat, Oct 30, 2010 at 6:22 PM, Brian Capouch bri...@palaver.net wrote: I wonder if anyone out there has a perspective on this. There are a welter of tickets out there on the matter, most of them closed. I'm actually able to reproduce this pretty often, for me using IAX2 with IMAP voicemail (google apps) is how. I haven't had much time to debug it, but plan to play more with it the coming weeks. Any update, Paul? /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feature Request for 1.10 - ISDN power-save mode
2 nov 2010 kl. 17.19 skrev Olivier: Hi, In Europe many Telcos implement power-save mode (See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to 'Activation / Deactivation' for more information). Would you agree to have this feature added to the ones already discuused for next Asterisk release ? (See https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2010) The projects you see on that list all have resources allocated to them or reasonable close to get allocated by the persons that participated in that meeting - unless you find them in the final categories (3.9 and 3.10). If you have development resources or funding and can create code that works, we are ALWAYS open for contributions, regardless of our lists. Looking forward to your contribution! Best regards, /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
