similar to: ulimit

Dear, do you have any successful experience for installing SHT-8C/PCI/FAX (synway) with asterisk ? is it compatibe with asterisk (dahdi/zaptel)? best -- Pezhman Lali -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110525/9df2050a/attachment.htm>

Dear, Faxter is an opensource email to fax gateway, please check it, let me know if any bug. best -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110130/0f418a92/attachment.htm>

Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 | 2 | hangup | | 455 | DID | 14193016880 | 1 | Dial | H323/1169#989181310524@66.152.61.66|60 | didx.org for

Hi, I would have thought that when spawning a channel using the Originate() dialplan command, variables prefixed with two underscores would be preserved. However this does not work in the following case. Dialplan code: [intern] exten => 200,1,Set(__myvar="foo") exten => 200,n,Originate(Local/123 at test_orig,exten,dummy) [test_orig] exten => 123,1,NoOp(${myvar}) exten =>

Dear this note is only for fresh administrators don't think about asterisk security. I found fail2ban very useful for anti asterisk hacking, so I want to share it with fresh admins. some hackers try your sip or iax2 ip with a lot of username/password, may be after 1 million try, one username/password was accepted. so in 2-3 hours, they use all of the credit of the hacked user. fail2ban, runs

Dear,is any way to find silent channels , and disconnect them after 30 secs? best -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090824/7709c910/attachment.htm

Hello On a uClinux-based appliance, "ps aux" shows multiple Asterisk processes: 380 root 11990 S asterisk -f 381 root 11990 S asterisk -f 383 root 11990 S asterisk -f 384 root 11990 S asterisk -f 385 root 11990 S asterisk -f 386 root 11990 S asterisk -f 387 root 11990 S asterisk -f 388 root 11990 S asterisk -f

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

Dear, Our iax clients's ip and port in the database were removed automatically, after 30 secs. the iax info is saved in odbc and postgresql . asterisk=# select * from iax_buddies where username='9706015'; name | username | type | secret | md5secret | dbsecret | transfer | inkeys | outkeys | auth | accountcode | amaflags | callerid | context | defaultip | host | language

hello is any web based sip phone? for example: a user after logining in, view a configured sip phone, and ...... best MAni ____________________________________________________________________________________ Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains. http://farechase.yahoo.com/promo-generic-14795097

Hi, On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest. Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly 1-2 concurrent calls. No other activity on server. Top shows asterisk on top. Its quad xeon server with 4 gb ram. Any suggestion where should I start and how should I go about with my investigation. Thank you and have a great weekend. Sans

Alex, Thank you so much for your response. I've been so consumed with other business that I only just now getting back to this issue. We have implemented your suggestion which is perfect. Thank you again. I've never asked a question of the community before and I'm extremely happy with the rapid response I received. Somewhat related to this initial problem I have an additional

Hi I have installed, asterisk , with postgresql. it 's the view of extensions table: didex=# select * from extensions order by id desc limit 5; id | context | exten | priority | app | appdata | description

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

Hi dears, I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) . I am facing problem with detecting caller id before first ring.I recorded the dahdi channel using dahdi_monitor command. Where I am able to see and hear caller-id dtmf tones. Pl tell me the procedure to upload recorded file if you needed. Something I want

Hy all of you, I've successfully installed a freepbx solution with 10 extensions : - 5 on Linksys SPA922 - 1 on Linksys SPA942 - 1 on Thomson ST022 Everything seems to work fine with all the hardphones excepts last week. The thomson has a strange behaviour. It can reach french mobile cell phones but when it reaches "fix" phones, the correspondant can't hear the caller. What

Dear I installed chan-sccp-b v3 on a powerful virtual machine, with 4 cpu cores and 16GB RAM(enabled in kernel by PAE) about 1,200+ clients are going to register in this machine. all data of clients are saved in ORACLE. The asterisk (1.6.2.18) connected to the database throw odbc(unixodbc). all logging are disabled( verbose, debug and sccp debug) . the asterisk was crashed every few minutes. here

Dear, Huawei softx3000 sends the dtmf with undefined content-type(sscc) and format, so the asterisk can not recognize the digits, maybe changing the source code of asterisk be a good solution, but I am looking for a better way. would you please let me know if you have a better solution. Best <--- SIP read from UDP:1.1.1.1:5060 ---> INFO sip:050111111 at 213.203.201.51:5060 SIP/2.0

Dears, There is a PSTN line shared between 2 asterisk servers, (openvox 4FXO lines) The outgoing call of the one server may be conflict with the established call of the other one, is any way to force the Asterisk or Dahdi to dial after hearing the Dial tone ? -- Pezhman Lali -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All; Why we use the agent password when we configure the agent in the agents.conf if the agent login by dialing the number configured in the extensions.conf? example: exten => 28, 1, AgentLogin(1001) I know that agent username is used to assign the agent to the queue, but when we use the agent password? Regards Bilal