Re: [asterisk-users] how to store sip.conf and extension.conf into phpmyadmin
It's clear in the log. no mysql engine found. I would suggest you to install unixodbc, configure res_odbc.conf and in extconfig.conf in place of mysql put odbc and a connection name defined in res_odbc.conf : Here's some snippets. odbc.ini --- [asterisk-connector] Description = A description here Trace = Off TraceFile = stderr Driver = MySQL SERVER = server_host PORT= 3306 DATABASE= database_name -- res_odbc.conf - [telephony] enabled = yes dsn = asterisk-connector username = db_username password = db_pass pre-connect = yes - extconfig.conf - sippeers = odbc,telephony,sipfriends sipregs = odbc,telephony,sipregs voicemail = odbc,telephony,voicemail_users extensions = odbc,telelephony,extensions queues = odbc,telephony,queue queue_members = odbc,telephony,queue_member followme = odbc,telephony,followme followme_numbers = odbc,telephony,followme_numbers Don't forget to create the appropriate tables and load res_odbc.so Also your shows that asterisk can't resolve the name sipauth.deltathree.com thus unable to register to it. Make sure that your server is connected to internet and configured with a DNS server. Have a good week-end. Adolphe Cher-Aime On Sat, Sep 13, 2014 at 12:35 PM, rafa alfurqan rafa.alfur...@gmail.com wrote: Hi, actually i just had trying Asterisk Full RealTime Database from this site http://blog.eduguru.in/tag/configure-asterisk-mysql-connection-create-the-res_mysql-conf-file-in-etcasterisk-vi-etcasteriskres_mysql-conf-enter-the-following-general-dbhost-127-0-0-1-dbname-asteriskrealtime-dbuser/ but in the end, i got failed. this is the log from CLI Connected to Asterisk 11.11.0 currently running on server-main (pid = 12017) [Sep 13 11:55:20] WARNING[12051]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x97ae2d8 (len 406) to (null) returned -1: Invalid argument [Sep 13 11:55:21] WARNING[12051]: config.c:2578 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available [Sep 13 11:55:21] WARNING[12051]: config.c:2578 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available [Sep 13 11:55:21] ERROR[12051]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo(sipauth.deltathree.com, (null), ...): No address associated with hostname [Sep 13 11:55:21] WARNING[12051]: acl.c:833 resolve_first: Unable to lookup 'sipauth.deltathree.com' [Sep 13 11:55:21] WARNING[12051]: config.c:2578 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available [Sep 13 11:55:21] WARNING[12051]: config.c:2578 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available [Sep 13 11:55:21] WARNING[12051]: acl.c:962 ast_ouraddrfor: Cannot connect [Sep 13 11:55:21] WARNING[12051]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x97ae2d8 (len 406) to (null) returned -1: Invalid argument [Sep 13 11:55:21] NOTICE[12051]: chan_sip.c:15218 sip_reg_timeout: -- Registration for '1212...@sipauth.deltathree.com' timed out, trying again (Attempt #596) [Sep 13 11:55:21] WARNING[12051]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x97ae2d8 (len 406) to (null) returned -1: Invalid argument [Sep 13 11:55:22] WARNING[12051]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x97ae2d8 (len 406) to (null) returned -1: Invalid argument actually, i use asterisk 11.11.0 and ubuntu 10.04 what should i do to solve that? thank you On 9/13/14, achera...@gmail.com achera...@gmail.com wrote: I would suggest you to use asterisk realtime. In this case your peers and extensions can be configured from database. https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration Sent from my iPhone On Sep 13, 2014, at 9:44 AM, rafa alfurqan rafa.alfur...@gmail.com wrote: hi, i want to sip.conf and extension.conf files could to import to the my database phpmyadmin, so the data that i had input to those file could be read into database? any help will be appreciated. thank you -- _ -- 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] High Availability with Asterisk
Good post. Actually this is the architecture we have. On Fri, Mar 7, 2014 at 11:31 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Thu, Mar 6, 2014 at 3:33 PM, Markus unive...@truemetal.org wrote: Hi Thorolf, Am 06.03.2014 16:21, schrieb Thorolf Godawa: Using (para-)virtualization with Xen could be an other option, on systems with low load this works reliable, but what happens on systems with high load? Are there any issues known about problems with the realtime, packet loss etc. because it runs in a VM? hmm, all my Asterisk'es run in (KVM) VMs, no issues there. But how is this related to high availability? I think it's not. :) I think the way to go for high availability (and scalability) is Kamailio! In a redundant setup, running on 2 separate physical machines (maybe in a VM, doesn't matter). Then you make them failsafe using whatever tool(s) available. Then you can set up 1, 2, 10 or 100 Asterisk behind Kamailio and any of them could fail (but 1 :-) ) and you will still be online. If you want to further develop the high availability thought, then you could use CephFS which will give you self-healing, 100% available storage over multiple physical storage servers. There you could store your Asterisk config files, or your MySQL database used by all the Asterisk servers, for CDRs, SIP registrations etc. It's kinda slow, but I think fast enough for Asterisk / MySQL. :) And, to scale and to make the Asterisk nodes redundant (redundancy is not really needed anymore, since Kamailio takes care of that, but basically then you get also VM/physical redundancy), you could look into OpenNebula which provides a nice auto-scaling feature already out of the box. If there's load on your Asterisk VMs, OpenNebula will detect this and spawn new Asterisk VMs (probably on different physical servers, otherwise it doesn't make that much sense performance-wise) which will automagically receive requests/calls from Kamailio. If the load goes down, the VM can be automagically stopped again to free resources for other VMs/applications. OpenNebula is less popular than OpenStack, which seems to be the first choice for Cloud-stuff today, but what I liked about OpenNebula is that it provides the auto-scaling feature already in the customer-facing web-frontend out-of-the-box, unlike OpenStack. So you could offer your customers a self-managed, redundant Asterisk cloud or something like that. :) In theory, this combination should give you a 100% redundant, auto-healing, auto-scaling VoIP setup. :) +1 to this post. A lot of good information here. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://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 -- _ -- 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] Starpy and Asterisk on different machines ?
Yes you can. This what starpy is for. It's build around Python twisted which allow you to write non blocked socket servers. You can use starpy as a fastagi server. Both AMI and FASTAGI can be configured from a .conf file as follow: [AMI] username=ami_user secret=ami_pass server=asterisk_ami_ip port=ami_port [FastAGI] port=listen_port interface=listen_ip Hope that will help. On Thu, Jan 16, 2014 at 10:02 AM, Olivier oza.4...@gmail.com wrote: Hello, Is it possible to run Starpy and Asterisk on different machines ? A quick glance at http://www.vrplumber.com/programming/starpy/ seems to tell it is possible but Debian's python-starpy package installs Asterisk. What do you think ? 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 -- _ -- 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] Web based Click to Call Application
Hi Marcus, You're right,WebRTC is the way to go. The only drawback is the fact that only astersik 11 support it natively. On Sat, Nov 10, 2012 at 3:41 PM, Markus Weiler markus_wei...@mailworks.orgwrote: Hi, I suppose WebRTC is the best solution nowadays, it's extremely interesting. I developed a C2C app in 2008, starting with call files and AMI, ended with asterisk-java and asterisk.NET to solve it. Hint: Try to solve (al)most (all) of your problems using Dialplans/Variables. Basically it's just one originate action using local channels. Markus Am 09.11.2012 11:38, schrieb Binan AL Halabi: Hi, Here is a starting point (WebRTC): https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support Regards. // Binan. -- *Från:* akhilesh chand omakhileshch...@gmail.comomakhileshch...@gmail.com *Till:* Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com asterisk-users@lists.digium.com *Skickat:* fredag, 9 november 2012 11:32 *Ämne:* [asterisk-users] Web based Click to Call Application Dear All, I want to develop click to call(C2C) web based application.Is there any study material. I will really appreciate your help, thank you. Regards Akhilesh -- _ -- 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 -- _ -- 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] dahdi 2.6.1+2.6.1 compile fails
Hi Eric, Make sure that you have the proper kernel and kernel-devel installed. Adolphe On Sat, Nov 3, 2012 at 4:32 PM, Eric Smith e...@fruitcom.com wrote: I am trying to compile a dahdi module from checkout: svn co http://svn.asterisk.org/svn/dahdi/linux-complete/tags/2.6.1+2.6.1 with ubuntu 3.5.0-17-generic and gcc 4.7.2 Error on compile is: oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_conf_bridge.c:3870:47:\ error: 'NULL' undeclared (first use in this function) This is identical to the error reported in this patch fix: https://issues.asterisk.org/jira/browse/DAHTOOL-60?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel How would I apply the patch included in the above url? [eric@pepper ~/src/asterisk-complete/asterisk/dahdi/2.6.1+2.6.1] $ patch DAHTOOL-60-f17.diff can't find file to patch at input line 5 Perhaps you should have used the -p or --strip option? The text leading up to this was: -- |diff --git a/xpp/oct612x/include/octdef.h b/xpp/oct612x/include/octdef.h |index a2da33d..7e534b7 100644 |--- a/xpp/oct612x/include/octdef.h |+++ b/xpp/oct612x/include/octdef.h -- File to patch: -- Eric Smith -- _ -- 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] Reliable SIP Trunk Provider
I use flowroute.com. Intuitive GUI, cheap, and good customer service. On Thu, Mar 15, 2012 at 1:30 PM, Guy Gold g...@the-golds.us wrote: On Thu,Mar 15 12:10:PM, Eric Wieling wrote: I'm a fan of Vitelity. They are no-frills, but they work well for my very low usage. I think their web portal is ugly, not all that intuitive, but it does work. I've been with them since early 2006 for my few low usage DIDs. +1 for Vitelity , I like them for recognizing the fact that some people actually prefer to run pure Asterisk (no GUI) . Guy Gold -- _ -- 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 -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- 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] number of calls simultaneous from AMI
Make sure that you set *async *option to true. If not asterisk will wait for response on previous calls before making any other calls. Hope that will help. On Wed, Sep 28, 2011 at 12:17 AM, Sam Govind govoi...@gmail.com wrote: If you can post any relevant code sections and CLI output for this then it'll be lot better to determine whats causing this. I never got any problem initiating as many call as u can say from AMI ! On Tue, Sep 27, 2011 at 5:36 PM, Jerry Geis ge...@pagestation.com wrote: I am starting up 4 calls over the AMI. It appears as though the first 3 start up and go out right away. The 4th call is delayed like 15 seconds. Any thoughts on why this fourth call might be getting delayed... Everything is working its just delayed. Jerry -- __**__**_ -- 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 -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macro to Dial a Channel Group using Round-robin
Try using local channel to accomplish that. by example : you have 2 phones in the group and want to dial those phones in the following fashion. Dial phone 1 first after 15 sec if phone1 does not pickup dial phone2 : [group-call] exten = group1,Dial(Local/phone1@group-callLocal/phone2@group-call, 30) exten = phone1,1,Dial(SIP/100,15) same = n,Hangup() exten = phone2,1,Wait(15) same = n,Dial(SIP/101) same = n,Hangup() Hope that will help. asterisk the definitive guide 2011/7/20 Antonio Modesto mode...@isimples.com.br ** Good morning, I am writing a Asterisk dialplan from scratch (for learning and testing purposes), but i'm having trouble with a algorithm to dial a SIP group using round-robin. I want that asterisk dial the member of the group in a circular way, until the call be answered. For example, i have the group TEST=SIP/1SIP/2SIP/3SIP/4, asterisk would dial SIP/1, if it doesn't answer in a period of time then asterisk would dial SIP/2 and so on. Can somebody help me? Thanks. -- _ -- 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 -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- 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 1.6.1 Realtime SIP Users
call-limit is deprecated in this version of asterisk. Use the callcounter and group count to limit calls On Thu, Jun 30, 2011 at 7:06 PM, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Hello, I just implement the SIP Peers with MySQL. In the structure mySQL missing the following fields: nat = yes notransfer = yes dtmfmode = rfc2833 call-limit = 2 canreinvite = no subscribecontext = blf subscribecontext (BLF) and call-limit (Protection) are very important ... Can you help me? Best, Mickael -- _ -- 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 -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- 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] Integration of OpenVXI
Check out this product. http://www.i6net.com On Mon, Jun 20, 2011 at 9:40 AM, Gopal krishnan gopalakrishnan...@gmail.com wrote: Hi, Have anybody integrated OpenVXI http://www.speech.cs.cmu.edu/openvxi/ with Asterisk? Thanks, Gopal -- _ -- 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 -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- 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] HA Asterisk
You can't do PRI failover while using internal PRI cards. To do so you need a standalone PRI box a good one i use often is foneBridge from Redfone. U can use foneBridge as follow: PSTN(Tdm) - FoneBridge(TDMoE)- Asterisk . Adolphe Cher-aime From my Iphone On Apr 30, 2011, at 9:09 AM, Satish Patel satish...@hotmail.com wrote: Tell me how to do pri failover. I meant we have one pri line but two asterisk in HA. Currently we are doing manually Swapping pri line. -- Sent from my iPhone On Apr 30, 2011, at 2:13 AM, RAJNIKANT VANZA rajniva...@gmail.com wrote: Hi Kaushal, I have done HA for Asterisk servers as well as SIP Server (kamailio). Please write your detail requirement. - how many Asterisk Sever require for HA? - How much down time acceptable during Asterisk Sever failover? - Which type Asterisk Sever Failover u required? Send me your detail requirement and answer of above question ASAP. -- Best Regards, Rajnikant Vanza Software Engineer --- Working On Linux,C/C++,VoIP,Asterisk Technology On Sat, Apr 30, 2011 at 7:59 AM, Kaushal Shriyan kaushalshri...@gmail.com wrote: Hi, I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf , but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA - High Availability.) Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? We have been evaluating Asterisk for our Voice Application and it seems it would fit the requirement. Is Asterisk a CPU Intensive or a Memory Intensive application. Please suggest/guide. Regards, Kaushal -- _ -- 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 -- _ -- 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] Call files or AMI originate for mass outbound call
Hello Guys, In the case of a multiserver environment for outbound automatic calls, can you share you experience and preference between call files and ami originate ? thanks -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- 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] Call files or AMI originate for mass outbound call
Thank you for your answer. I also prefer AMI for its flexibility. However, i have an application developped in PHP used to make more than 10 calls a day by group of 120 concurrent calls. My problem with AMI is that client keeps disconnected to AMI server. I use astmanproxy as proxy server. Do you to use have such problem with your applications ? Regards On Wed, Apr 20, 2011 at 3:11 PM, Paul Belanger pabelan...@digium.comwrote: On 11-04-20 12:20 PM, Adolphe Cher-Aime wrote: In the case of a multiserver environment for outbound automatic calls, can you share you experience and preference between call files and ami originate ? I prefer using the AMI as I have better call control. I also get to monitor the AMI events are react to them. Recently I've been using Python and starpy (twisted). -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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 -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- 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] Call files or AMI originate for mass outbound call
Thanks Paul, I will take a look at twisted i will let you know. Regards On Wed, Apr 20, 2011 at 5:38 PM, Paul Belanger pabelan...@digium.comwrote: On 11-04-20 04:44 PM, Adolphe Cher-Aime wrote: Thank you for your answer. I also prefer AMI for its flexibility. However, i have an application developped in PHP used to make more than 10 calls a day by group of 120 concurrent calls. My problem with AMI is that client keeps disconnected to AMI server. I use astmanproxy as proxy server. Do you to use have such problem with your applications ? Not really, twisted (specifically the ClientFactory) has functions to handle disconnection / reconnection. It is transparent to my application, so if the client is disconnected from Asterisk, and event is fired, I stop processing calls, then wait for the client to reconnect to the AMI. Once reconnected, I begin again. [1] http://twistedmatrix.com/documents/current/core/howto/clients.html#auto4 -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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 -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- 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] Realtime SIP peer status
Registry type Event will give you information about your peer. Adolphe Cher-aime From my Iphone On Apr 15, 2011, at 1:15 AM, Jonas Kellens jonas.kell...@telenet.be wrote: On 04/13/2011 09:18 PM, Rob Coward wrote: Rather than add extra overhead to your dialplan and the asterisk server, why not make use of the AMI and have a background process listening for the various events and updating your database accordingly ? See http://www.voip-info.org/wiki/view/asterisk+manager+events#ExtensionStatusEvent and http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.5/lib/Asterisk/AMI.pm#Events Regards, Rob Hello, this event tells me something about an extension, but not about the SIP peer status. Kind regards, 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 -- _ -- 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] Duplicate cdr records with channel local
Hi list, I use AMI to originate calls for outbound campaigns via local channel. I use CDR records to have status of the calls such as : disposition, if call was answered by live human or answering machine, reason of call failure and make accounting based on total calls that have been made. My problem is that I have more than one CDR records for a single call. One for the Leg A and another one for Leg B. As i use cdr manager to retrieve call statistics of the campaign duplicate CDR records is really an issue. Can anybody please help me on this ? Or is there a better way to have those stats. I use asterisk trunk 1.6.2 . Your help is greatly appreciated. LIVE Every Moment, LOVE Every Day Let the nature do the rest and so NERVER GIVEUP !!! -- _ -- 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] GotoIf CALLERID(num)
I don't think you need the quotes. Try without them
Adolphe Cher-aime
From my Iphone
On Dec 29, 2010, at 6:41 PM, Joseph syscon...@gmail.com wrote:
No, it is not a space issue, I tried:
exten = s,3,GotoIf($[${CALLERID(num)}=4715665]?4:6)
but it still goes to priority 6
--
Joseph
On 12/29/10 16:23, Joel Maslak wrote:
Get rid of the spaces before and after the equal sign.
On Wed, Dec 29, 2010 at 4:15 PM, Joseph syscon...@gmail.com wrote:
I'm testing GotoIf($[${CALLERID(num) but I'm missing something as
it is not
working:
[office-open]
exten = s,1,Wait(1)
exten = s,2,Answer()
; for Caller ID is 471-5665, always signal congestion:
exten = s,3,GotoIf($[${CALLERID(num)} = 4715665]?4:6)
exten = s,4,Playtones(congestion)
exten = s,5,Congestion(5)
exten = s,6,SetMusicOnHold(default)
...
but it always goes to s,6
What am I missing?
--
Joseph
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Re: [asterisk-users] start services automatically
When installing asterisk you should type make config to have asterisk create init script automatically. For http chkconfig httpd on Adolphe Cher-aime From my Iphone On Dec 20, 2010, at 5:48 AM, salaheddine elharit salah.elharit...@gmail.com wrote: Hello All, i have asterisk installed in my call centre without any issue I would like to ask you some questions related to services. i want to start asterisk and httpd and aheevacti automatically when the server centos reboot or shutdown becouse i must start all services manually (service asterisk start ,service httpd start ...) Maybe i must use crontab but I don’t know how to do, any help please Thanks and 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 -- _ -- 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 dynamic span error
Hi Everybody, I'm trying to connect an asterisk box to a provider using Redfone Fonebridge dual E1. Installation seems to run correctly only i can't get the D Channel up and i have the following error displayed. DYN/ SPAN ethmf mac_address_fo_fbport Expected seq no 0 , but received 3456 instead . This error keep scrolling until i disconnect the ethernet cable connected to the FoneBridge. Any Help please. -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- 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] Get the Uniqueid of Action Originate in the AMI
Set event on while login into AMI and set your own uniqueid using action ID for that call . Example : action: login Username: your_user Secret: your_secret Event: On action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr ActionId: yourID Hope that will help. On Mon, Nov 8, 2010 at 1:12 PM, Rodrigo Lang rodrigoferreiral...@gmail.comwrote: Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a lots, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- 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 -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- 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 Redundancy
Hello List, I need a weed to load balance some asterisk boxes that have pstn connectivity via E1. The problem is that i will not use sip phones but instead call files for auto dialing. Is is possible to load balance when call are generated from call files? Thank you so much. On Sun, Sep 26, 2010 at 10:31 AM, Michelle Dupuis mdup...@ocg.ca wrote: Check out HAAST (High Availability ASTerisk) at www.generationd.com (also on the voip wiki) You get the cluster/heartbeat replication without needing to add openSER or full HAlinux. A simpler approach - easier to config and manage MD -- *From:* asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo [ d...@keshercommunications.com] *Sent:* Sunday, September 26, 2010 11:04 AM *To:* Asterisk Users List *Subject:* [asterisk-users] Asterisk Redundancy Hello, Are there any guides to setting up high-availability asterisk platforms? Maybe using Opensips. I found this diagram, but i cant find any guides on how to go about setting it up. http://yfrog.com/5unetworkexampleg Thanks Dan -- _ -- 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 -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- 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] Unable to make outgoing call on E1
Hello everyone, I'm using redfone fonebridge to have Pstn connectivity on my asterisk box. I can receive in coming calls however outgoing calls don't go to provider. It's seems it's a span config problem. Because in systemconf when I try to config span as follow span=1,0,2,ccs,hdb3,crc4 And start dhadi I have the following message Unable to config ... Bad argument(22). Please any help Adolphe Cher-aime From my Iphone -- _ -- 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] Max TDM calls per asterisk box
Hi Everyone, Can you tell me how many concurrent TDM (Dahdi) calls that a single asterisk box can handle. Configuration is as follow : Quad core Xeon 3 GHZ, 4Gb RAM, asterisk 1.6.2.9 Also do you know a good tool to stress out asterisk? Kind regards -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- 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] Max TDM calls per asterisk box
Thank you for your quick reply. I'll use Redfone foneBridge to terminate E1s. I won't use any transcoding. For now I opt for software based echo cancellation. Thanks. On Wed, Sep 8, 2010 at 6:43 PM, Paul Belanger paul.belan...@polybeacon.comwrote: On Wed, Sep 8, 2010 at 5:57 PM, Adolphe Cher-Aime achera...@gmail.com wrote: Quad core Xeon 3 GHZ, 4Gb RAM, asterisk 1.6.2.9 How many PCI / PCI-X slots do you have? Do you plan to use hardware Echo Cancellation? What about transcoding? Theses will be your basic concerns when dealing with concurrent calls. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- 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] Configuring X-lite for a remote user
To have your asterisk box reachable from internet you must configure static nat on your router to get sip traffic to the public Ip redirected to your internal ip. Make sure that sip and rtp traffic are not bloked by firewall. And configure xlite to connect to your public ip address. Adolphe Cher-aime From my Iphone On Jul 26, 2010, at 7:48 PM, ayodele abejide ayodeleabej...@hotmail.com wrote: I have asterisk running at home, a friend would be traveling out of the country and I want him to be able to put a call through from his remote location, I am wondering how I would configure the X-lite client on his pc so he would be able to call through assuming my public address is A.B.C.D and the static address the asterisk machine is on is 192.168.0.3. Thanks in anticipation Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Si gn up now. -- _ -- 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] Video IVR Asterisk ?
Anita, It's possible to do so. With I6NET Vxml sofware and some addons. Check out http://www.i6net.com Adolphe Cher-aime From my Iphone On Jul 16, 2010, at 1:38 PM, Anita Hall anita.h...@simmortel.com wrote: Hi Is it possible to receive video calls using Asterisk and then process them as an IVR ? One of our clients wants to set-up a video IVR system in the US and we are evaluation possible options. Also, what is the bandwidth of receiving a video call in US ? What protocols and codecs are supported and does it work on DID numbers ? Can I rent a hosted solution for this ? Thanks in anticipation of your valuable inputs. regards, Anita Hall, Simmortel. -- _ -- 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] g729 codec loading
Hello Everyone, I've successfully registered my g729a licenses. When i try to load the module from asterisk Cli i got the following error *Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied* * loader.c:795 load_resource: Module 'codec_g729a.so' could not be loaded.* * * *I'm running asterisk 1.6.2.9 on CentOs 5.4 * * * *Best regards* * * -- Adolphe CHER-AIME Network Integrator CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3748-3875 / (509) 3449-4280 -- _ -- 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] g729 codec loading
Thank you Moises it works perfectly. On Fri, Jul 16, 2010 at 5:07 PM, Moises Silva moises.si...@gmail.comwrote: Try disabling SELinux if you have it enabled (unless of course you need it). I seem to remember there is certain compilation flags required (position independent code, -fPIC?) to run with SELinux enabled, may be the codec_g729a.so is not compiled properly to run under such circumstances? Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com On Fri, Jul 16, 2010 at 5:57 PM, Adolphe Cher-Aime achera...@gmail.comwrote: Hello Everyone, I've successfully registered my g729a licenses. When i try to load the module from asterisk Cli i got the following error *Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied* * loader.c:795 load_resource: Module 'codec_g729a.so' could not be loaded.* * * *I'm running asterisk 1.6.2.9 on CentOs 5.4 * * * *Best regards* * * -- Adolphe CHER-AIME Network Integrator CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3748-3875 / (509) 3449-4280 -- _ -- 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 -- Adolphe CHER-AIME Network Integrator CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3748-3875 / (509) 3449-4280 -- _ -- 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] Assign dhadi channel to several groups
Hello guys, I was wondering if it's possible to assign a dahdi channel to two diferent groups. Thanks Adolphe Cher-aime From my Iphone -- _ -- 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] Assign dhadi channel to several groups
Thank you Tzafrir Adolphe Cher-aime From my Iphone On Jun 6, 2010, at 11:28 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Jun 06, 2010 at 11:27:45AM -0500, Adolphe Cher-aime wrote: Hello guys, I was wondering if it's possible to assign a dahdi channel to two diferent groups. Sure. No problem: group = 1,2,3,5,8,13,21,34,55 channel = 15 -- 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 -- 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] Asterisk Cluster
Thank you David. I was thing about the cisco solution but cost is the issue as I will so many DSP to for this amount of calls. Regards Adolphe Cher-aime From my Iphone On May 19, 2010, at 4:23 PM, David Backeberg dbackeb...@gmail.com wrote: On Wed, May 19, 2010 at 12:38 AM, Adolphe Cher-Aime achera...@gmail.com wrote: Hello Everyone, I must deploy an asterisk system that can support at least 500 pstn outbound calls. It's a challenge as it's the first time i'm gonna build such a large system. I want to have your advice on hardware, software and so on . What i have in my plan is a cluster of servers with quad PRI cards. I will appreciate your advice. I don't know what you've done for your smaller deployments. Different people have different opinions. My personal preference, if this is United States: DS3 - Adtran channel bank, break out PRIs PRIs - multiple Cisco PRI-SIP appliances. A 3845 maxed out can do 24 PRIs. I recommend using two and splitting load for capacity / failover. Using pure SIP between the gateway appliances and at least two asterisk boxes for flexibility and failover-ability. -- _ -- 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] Asterisk Cluster
Thank you Jonathan. I really appreciate Adolphe Cher-aime From my Iphone On May 19, 2010, at 6:26 PM, Jonathan Thurman jonat...@thurmantech.com wrote: On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime achera...@gmail.com wrote: Hello Everyone, I must deploy an asterisk system that can support at least 500 pstn outbound calls. It's a challenge as it's the first time i'm gonna build such a large system. I want to have your advice on hardware, software and so on . What i have in my plan is a cluster of servers with quad PRI cards. I will appreciate your advice. Your up front cost is going to be a little higher with TDM - SIP devices, but your management will be a lot easier. AudioCodes also has equipment that can support a DS3 connection, or multiple T1s directly to SIP. For example, If you are getting 22 T1s then get two AudioCodes Mediant 2000s with 16 spans (maxed out) or two Mediant 3000s with 21 spans. If you are getting a DS3, get a Mediant 3000. The M3000 supports up to 84 T1s or three DS3 or one OC3. Then leave call management up to Asterisk. Of course, have redundancy everywhere you can. -Jonathan -- _ -- 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] Asterisk Cluster
Jonathan for redundancy which software do you recomand? Adolphe Cher-aime From my Iphone On May 19, 2010, at 6:26 PM, Jonathan Thurman jonat...@thurmantech.com wrote: On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime achera...@gmail.com wrote: Hello Everyone, I must deploy an asterisk system that can support at least 500 pstn outbound calls. It's a challenge as it's the first time i'm gonna build such a large system. I want to have your advice on hardware, software and so on . What i have in my plan is a cluster of servers with quad PRI cards. I will appreciate your advice. Your up front cost is going to be a little higher with TDM - SIP devices, but your management will be a lot easier. AudioCodes also has equipment that can support a DS3 connection, or multiple T1s directly to SIP. For example, If you are getting 22 T1s then get two AudioCodes Mediant 2000s with 16 spans (maxed out) or two Mediant 3000s with 21 spans. If you are getting a DS3, get a Mediant 3000. The M3000 supports up to 84 T1s or three DS3 or one OC3. Then leave call management up to Asterisk. Of course, have redundancy everywhere you can. -Jonathan -- _ -- 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] Asterisk Cluster
Thanks Michelle Adolphe Cher-aime From my Iphone On May 19, 2010, at 8:02 PM, Michelle Dupuis mdup...@ocg.ca wrote: The High Availability HASTerisk (HAAST) product on www.generationd.com is a software solution that does automatic failover, etc between multiple asterisk machines. I'm guessing this could be part of an overall solution for you From: asterisk-users-boun...@lists.digium.com [asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff Brower [jbro...@signalogic.com] Sent: Wednesday, May 19, 2010 8:02 PM To: Jonathan Thurman Cc: Asterisk Users List Subject: Re: [asterisk-users] Asterisk Cluster Jonathan- On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime achera...@gmail.com wrote: Hello Everyone, I must deploy an asterisk system that can support at least 500 pstn outbound calls. It's a challenge as it's the first time i'm gonna build such a large system. I want to have your advice on hardware, software and so on . What i have in my plan is a cluster of servers with quad PRI cards. I will appreciate your advice. Your up front cost is going to be a little higher with TDM - SIP devices, but your management will be a lot easier. AudioCodes also has equipment that can support a DS3 connection, or multiple T1s directly to SIP. For example, If you are getting 22 T1s then get two AudioCodes Mediant 2000s with 16 spans (maxed out) or two Mediant 3000s with 21 spans. If you are getting a DS3, get a Mediant 3000. The M3000 supports up to 84 T1s or three DS3 or one OC3. Then leave call management up to Asterisk. Of course, have redundancy everywhere you can. In this scenario, does Asterisk still touch every RTP packet? Or can 'native bridge' mode be enabled? -Jeff -- _ -- 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 -- _ -- 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 Cluster
Hello Everyone, I must deploy an asterisk system that can support at least 500 pstn outbound calls. It's a challenge as it's the first time i'm gonna build such a large system. I want to have your advice on hardware, software and so on . What i have in my plan is a cluster of servers with quad PRI cards. I will appreciate your advice. Thank you all . -- Adolphe CHER-AIME Network Integrator CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3748-3875 / (509) 3449-4280 -- _ -- 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] OK, I'm stumped
Mi too I've experienced the same problem with my script. Dahdi answers the channel once Ami is running it's the same thing for call files . When using sip Chanel or skype channel it work as I wanted. I thank that analog fxo is the problem if automatic outgoing calls when you want the called party to answer first befor moving to the context extension. Adolphe Cher-aime From my Iphone On May 16, 2010, at 1:32 PM, Jose Flores Galicia floj...@gmail.com wrote: Maybe because I am closer to several customers which often make questions like yours. I can supposse you mean that the call is answered by the dahdi channel as soon as you set the originate command on AMI, I supposse you are using an FXO channel connected to your POTS line. Am I right? Jose Flores Galicia floj...@gmail.com BriefCode Code Based Training 2010/5/16 Bruce Ferrell bferr...@baywinds.org I'm trying to make an AMI call. I want to call a number, play an announcement when the call is answered, then call a second number and connect the two when the second call is answered. I an able to make a simple call to two numbers and connect them using the manager API but playing the announcement has me beat. Suggestions anyone? Bruce Ferrell -- _ -- 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 -- _ -- 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] Problem of Playing 'pbx-transfer'
Try x-lite to x-lite, snom to snom . That may be a codec problem. Which codec are you using? Adolphe Cher-aime From my Iphone On May 9, 2010, at 11:11 PM, Dovid Bender asteriskus...@dovid.net wrote: Process of elemination. Test with multiple phones, check the codec being used and make sure the file is there and available. - Original Message - From: kamrun nahar bina To: asterisk-users@lists.digium.com Sent: Friday, May 07, 2010 07:33 Subject: [asterisk-users] Problem of Playing 'pbx-transfer' Dear all, We have been using asterisk for 4 years. Now we have got problems which occurs during the attended transfer. During attended transfer, sometimes we cannot hear the sound of 'pbx- transfer'. I cannot understand why this is happening? log is : -- Started music on hold, class 'default', on SIP/113.34.235.13- b7a3f110 -- SIP/185148-092db338 Playing 'pbx-transfer' (language 'jp') Although it is showing Playing 'pbx-transfer' (language 'jp'), but it cannot hear 'pbx-transfer' sound Sometimes we can hear the sound of 'pbx-transfer'. is it the problem of network load or phone-set or something else? Please let me know. I am using x-lite and snom 300. Before i tested it for memory load, And found out that it is not a memory problem. Our system is as like as: The number of User agent is: 1650 The number of Actual registered user agent is: 600 Our System configuration is : IBM X3550 CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz HDD: 3.5 SATA 1TB x 2 version of asterisk: 1.4.23.1 our memory size is 4GB. concurrent calls no : 30. Our memory condition is below : Cpu(s): 0.3%us, 0.7%sy, 0.0%ni, 98.5%id, 0.0%wa, 0.1%hi, 0.3%si, 0.0%st Mem: 4147888k total, 3986540k used, 161348k free,76852k buffers Swap: 2031608k total, 56k used, 2031552k free, 3170396k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 23160 root 15 0 440m 415m 5688 S 4.3 10.3 398:13.93 asterisk Our disk space condition is below: FilesystemSize Used Avail Use% Mounted on /dev/mapper/VolGroup00-LogVol00 901G 245G 610G 29% / /dev/sda1 99M 18M 77M 19% /boot tmpfs 2.0G 0 2.0G 0% /dev/shm Asterisk and the User-Agent is connected through the Internet. ..And Is there any solution to solve this problem? I have investigated in several places but I cannot find out the reason? I need this solution very urgently. Is there any one who can solve this problem? -- _ -- 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] automatic call with call files
Hello asterisk gurus, I'm developping a script that create call files dynamicly from a database. Here the scenario script move call file to outgoing dir to place the call call is connected to [extension] which contains a playback app.While line is ringing, playback is triggered I want to start playback file when call is answered to make sure that called party hears message from beginning. What's is different now. I try with Wait and WaitForSilence, nothing is fixed. Does anybody know how to fix that. Your help would be highly appreciated. -- -- Adolphe CHER-AIME Network Integrator CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3748-3875 / (509) 3449-4280 -- _ -- 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
