Re: [asterisk-users] asterisk realtime database structure
On 8/4/12 10:38 AM, virendra bhati wrote: best link for asterisk realtime is below one http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example On Fri, Aug 3, 2012 at 1:51 PM, Leandro Dardini ldard...@gmail.com mailto:ldard...@gmail.com wrote: If you check the contrib/realtime/mysql directory in the source tree, you'll find scripts for almost all the tables. Thank you all for the hints! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw -- _ -- 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.ctl file
Hello guys, i've a little question to ask. What is the file asterisk.ctl ? Thanks, Regards. -- _ -- 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] SIP register refresh time
Hi all, question about register refresh time. One of our supplier had a maintenance work on sat 4 Aug which was replacing the production server for an Asterisk 1.4 running version. We have few Asterisk connected to them (version 1.6, 1.8 and 1.10) with register Username and Passwd. After the new server came up, no one of our Asterisks get registered back, which means no calls -incoming and outgoing- at all since this date :-( A sip show registry this morning (Mon 6 Aug) show us following: Hostdnsmgr Username Refresh State Reg.Time sip.domain.com:5060 N MyUser105 No Authentication Sat,04 Aug 2012 16:55:37 A simple sip reload made thinks working again. Why our Asterisks didn't get back for registration, refresh register time being the standard 120 seconds? Thanks for your explanation -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] - SIP retransmission problem
Hi Paolo, I had yesterday a similar problem and it was caused by a misconfigured IP address in extensions.conf that I forgot to update after changing some IP addresses in my network. Check the network connectivity between you Asterisk host and 1000. Double check that the IP address is correct. Use tcpdump to see what's going on the wires. Good luck! -- Jorge Martínez López jorg...@gmail.com http://www.jorgeml.net -- _ -- 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 tls problem
On 06/08/12 02:59, Vladimir Mikhelson wrote: Have you tried 1.8.15? I'm trying 1.8.13 because that is the versions currently scheduled for release in Debian 7 (wheezy) http://packages.debian.org/wheezy/asterisk If 1.8.15 contains definite solutions for TLS problems, then either a) they can be applied as patches on the Debian package of 1.8.13 b) there could be some attempt to get 1.8.15 accepted into Debian (the catalog for wheezy is technically frozen now for final testing before release, so they are not keen to accept whole new versions of packages) SIP TLS with self-signed certificate seems to be working fine here. The OS is CentOS 5.8 and there are no chained certificates in my environment. -Vladimir The original poster was also using self-signed certs I've observed the problem using chained certs (with 1 root, 2 intermediate, and then my server cert) -- _ -- 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] Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration Tutorial
Hello, I released an update to my series of Kamailio and Asterisk Realtime Integration, using the latest stable versions of the two projects, respectively 3.3.1 and 10.7.0. You can find it at: * http://asipto.com/u/68 The tutorial focuses on how to use Asterisk's database structure to perform authentication in Kamailio SIP server, along with user location, nat traversal, instant messaging, presence, a.s.o., offloading processing from Asterisk. Asterisk will still handle all the calls, enabling rich telephony such as MoH, transcoding, ring back, IVR, etc. Reusing as much as possible the Asterisk database makes the architecture presented in the tutorial easy to be applied to existing installations, without losing management interfaces or other admin tools. Hope it is useful for many folks out there. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw -- _ -- 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.ctl file
On Mon, Aug 06, 2012 at 10:03:41AM +0200, Giuseppe Longo wrote: Hello guys, i've a little question to ask. What is the file asterisk.ctl ? That is a UNIX Domain Socket file used to pass commands to an Asterisk process. It's how asterisk -r and asterisk -rx communicate with the back-end process that is doing the work. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration Tutorial
Thats a great tutorial with very good conceptual details like SIP messages flow. Thanks Daniel :) On Mon, Aug 6, 2012 at 6:48 PM, Daniel-Constantin Mierla mico...@gmail.comwrote: Hello, I released an update to my series of Kamailio and Asterisk Realtime Integration, using the latest stable versions of the two projects, respectively 3.3.1 and 10.7.0. You can find it at: * http://asipto.com/u/68 The tutorial focuses on how to use Asterisk's database structure to perform authentication in Kamailio SIP server, along with user location, nat traversal, instant messaging, presence, a.s.o., offloading processing from Asterisk. Asterisk will still handle all the calls, enabling rich telephony such as MoH, transcoding, ring back, IVR, etc. Reusing as much as possible the Asterisk database makes the architecture presented in the tutorial easy to be applied to existing installations, without losing management interfaces or other admin tools. Hope it is useful for many folks out there. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggestion of Server Specifications for Asterisk
I have bought a new server today: i7-2600 CPU, 8GB and 2 x 256GB SSDs. 100Mbit Connection. I hope CPU is powerful enough for 200 concurrent calls. On Sun, Aug 5, 2012 at 1:57 AM, Michelle Dupuis mdup...@ocg.ca wrote: That's how we do it - write to a memory based (ramdisk) disk then write to HDD upon call completion. We haven't tried a SSD but that may be necessary depending on your call volumes. -- *From:* asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner [ rswago...@gmail.com] *Sent:* Saturday, August 04, 2012 7:34 PM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Suggestion of Server Specifications for Asterisk On Sat, Aug 4, 2012 at 1:22 PM, Shahid H shah...@gmail.com wrote: Instead of buying expensive disk.. I might setup a ramdisk (about 2GB) to do 200 calls recordings. Once the call hangup/completed it will then move recording file to SATA HDD. What do you think of this? You want some form of raid for redundancy. I usually go with two 15K SAS drives in raid 1 or four 7.2k SATA drives in raid 10. Performance between the two should be similar. With drives being as cheap as they are skip raid 5. Ryan -- _ -- 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 1.6 and Outbound SIP Proxy configuration
Hello, Using asterisk 1.6 as sip client to register with sip provider and terminate calls through them. SIP Provider has provided sip proxy and sip server details. The problem is that the sip server FQDN does not resolve on the internet. So I can only presume that the SIP proxy knows how to reach the sip server. Asterisk 1.6 seems to have a problem with this. This is my config below: -- [trunk1] defaultuser=x...@sip.provider.com fromuser= fromdomain=sip.provider.com type=peer secret=a outboundproxy=10.10.10.10 ;(replaced actual ip) nat=no host=sip.provider.com dtmfmode=auto disallow=all context=from-internal canreinvite=no allow=g729 trustrpid=yes sendrpid=yes register = x...@sip.provider.com:a@10.10.10.10:5060 -- With the above config, I can register with the providers sip proxy, however, the error below is observed in the logs concerning the host when I try to make a call: -- [2012-08-02 23:37:31] WARNING[26155] acl.c: Unable to lookup ' sip.provider.com' [2012-08-02 23:37:31] ERROR[26155] chan_sip.c: srvlookup failed for host: sip.provider.com, on peer trunk1, removing peer -- I have done some research on this issue but not been able to find anything conclusive on why this would happen. I tested the sip details provided with a different sip client (actually an IP phone) and was able to register and send / receive calls with no problem. The problem just seems to be somewhere in my asterisk client configuration or a known bug with the version of asterisk I am using for this. Any pointers? Thanks. Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Block outbound calls based on IP address
We are looking to further secure our Asterisk installation by inspecting the IP address that a SIP INVITE comes from and performing some logic to determine whether the call should proceed. The purpose of this is to prevent calls to certain expensive destinations if the SIP message is coming from a foreign IP that we don't expect. I can see that it's possible to use the SIP_HEADER function however that may not contain the public IP address. For example here is an invite from the external IP address 58.28.1.1 but that information is not contained in the SIP header: U 58.28.1.1:5060 - 203.89.1.1:5060 INVITE sip:1...@domain.com SIP/2.0..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-fc116e03a80ef774-1---d8754z-;rport. .Max-Forwards: 70 ..Contact: sip:000333082261336@192.168.1.103:5060..To: sip:1...@domain.com..From: sip:000333082261...@domain.com;tag=7 dcb1e4d..Call-ID: NDMyZmRhY2Q4ZjNhMjAxMDJhOTA3OTU0MzMyNTkzNjI...CSeq: 1 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF O..Content-Type: application/sdp..Supported: replaces..User-Agent: X-Lite release 5.0.0 stamp 67284..Content-Length: 217v=0..o=- 12988751314362048 1 IN IP4 192.168.1.103..s=CounterPath X-Lite 5.0.0..c=IN IP4 192.168.1.103..b=AS:1638..t=0 0..m=audio 5062 RTP/AVP 0 8 3 101..a=rtpmap:101 telephone-event/8000..a=fmtp:1 01 0-15..a=sendrecv.. Is it possible to determine the public IP address from the dialplan? Any advice appreciated. attachment: winmail.dat-- _ -- 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] Background, Playback wave files in asterisk
Dears; I discover that I have to place the wave files in the /var/lib/asterisk/sounds/custom/ So, can I understand that the only solution I have is to copy the files that are existed in the path /var/lib/asterisk/sounds/en/ to the path /var/lib/asterisk/sounds/custom? Or there is any other solution? I am using FreePBX and the asterisk version is: Asterisk 1.8.11-cert1 Any advise? Regards Bilal - Hello; What is the difference between using the Background Playback in Asterisk 1.8 without cert and Asterisk 1.8 cert? I surprised that in cert version, I do not hear the sound ! And it is not working properly, but in the normal version, it is working. So what is the new? Is it the version? Or there are some variables or settings need to be done in asterisk 1.8 cert that was not require in the normal version (not cert)? Regards Bilal -- _ -- 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] Showing the name of the called number at the source IP Phone, how?
Hi All; Asterisk 1.8.11-cert1 I need to do the following, how? If my extension is 500 and I need to call the extension 501, so when dialing 501, then I need to be able to see the name of the 501 (for example, the name was: Mike, so I need to see at my IP Phone that I am calling Mike which is the name of the destination). How? Regards Bilal -- _ -- 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] Showing the name of the called number at the source IP Phone, how?
Hi, You need to set rpid on the calling phone settings, if that phone knows what to do with RPID. Then you need to set allowrpid=yes in the sip peer settings of A party and B party. I did that on CISCO 79X0 phones and it worked perfectly, Regards, Sammy On Tue, Aug 7, 2012 at 3:43 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; Asterisk 1.8.11-cert1 I need to do the following, how? If my extension is 500 and I need to call the extension 501, so when dialing 501, then I need to be able to see the name of the 501 (for example, the name was: Mike, so I need to see at my IP Phone that I am calling Mike which is the name of the destination). How? Regards Bilal -- _ -- 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] Showing the name of the called number at the source IP Phone, how?
www.voip-info.org/wiki/view/Asterisk+multi-language -- Best regards, Rudi -Original Message- From: bilal ghayyad bilmar...@yahoo.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 6 Aug 2012 15:43:24 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Showing the name of the called number at the source IP Phone, how? Hi All; Asterisk 1.8.11-cert1 I need to do the following, how? If my extension is 500 and I need to call the extension 501, so when dialing 501, then I need to be able to see the name of the 501 (for example, the name was: Mike, so I need to see at my IP Phone that I am calling Mike which is the name of the destination). How? Regards Bilal -- _ -- 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] Showing the name of the called number at the source IP Phone, how?
It's look like I'm wrong, didn't read your reply first, don't know there's such feature, very nice info :D -- Best regards, Rudi -Original Message- From: SamyGo govoi...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 7 Aug 2012 09:24:19 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Showing the name of the called number at the source IP Phone, how? -- _ -- 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
