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Dear All, I have make a queue in my dailplan and queue is not working properly,prbolem is that all call goes to same extenstion at a time.Because,I use eyeBeam(softphone) and eyeBeam have six line and whenever a call comes into eyeBeam that call reserved by Line 1 suppose to 2nd call will come that call goes to Line 2(same extension used by Line 1) and 3rd call goes to 3rd line and so on. But i

Hi folks, I'm not able to install asterisk whenever I hit make command I get below error: make[1]: *** No rule to make target `../main/modules.link', needed by `asterisk'. Stop. make: *** [main] Error 2 Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

HI folks, I have set a barging facility with our production box.Client able to barge a agent but client raise a requirement, they want talk to barge agent but that communication is not listen by customer. It is possible with asterisk or not. thanks in advance. Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Greg, I moved REJECT rule to last in the list but I'm getting same error. Regards Akhilesh On Mon, Apr 20, 2015 at 5:57 PM, Greg Woods <greg at gregandeva.net> wrote: > > On Mon, Apr 20, 2015 at 1:58 AM, akhilesh chand <omakhileshchand at gmail.com > > wrote: > >> Chain INPUT (policy ACCEPT) >> target prot opt source destination >> ACCEPT all -- 0.0.0.0/0 0.0.0.0/0 state >> RELATED,ESTABLISHED >> ACCEPT icmp -- 0.0.0.0/0 0.0.0.0/0...

hello all, i have two asterisk server for call transfer and one more asterisk server for agent login(server_X) where agent take the call. server_A and server_B server_A is connected with pri and configure with 60 channel for call transfer into server_X server_B is connected with pri and configure with 30 channel for call transfer into server_X my query is that some time two call originate same

...ve implemented several ec2 instances with both sip and iax2 and have > no problems with xlite or hard phones. Have you already opened the ports in > the vpc security group on the Amazon side? Let me know is I can help. > --James > On Apr 24, 2015 3:34 AM, "akhilesh chand" <omakhileshchand at gmail.com> > wrote: > >> Hi Thomas, >> >> Could you tell how can I change the protocol of corresponding port means >> 5060 is configured with tcp protocol I want to configured with udp. When I >> execute nmap -p5060 xx.xx.xx.xx I got below output >> &...

Hi Thomas, I followed your recommended command in asterisk CLI which is mentioned in above chain mail but I'm not able capture any log related to softphone. Chain INPUT (policy ACCEPT) target prot opt source destination ACCEPT all -- 0.0.0.0/0 0.0.0.0/0 state RELATED,ESTABLISHED ACCEPT icmp -- 0.0.0.0/0 0.0.0.0/0 ACCEPT all

Hi Folks, I'm trying to register softphone(X-lite) but I'm not able to register softphone whenever I'm trying to register softphone I got below error [image: Inline image 1] Is there any document/guide line where I will get process to register softphone in asterisk(Which is installed in EC2 Cloud). Regards Akhilesh -------------- next part -------------- An HTML attachment was

yes I called On Mon, Apr 13, 2015 at 1:27 PM, jg <webaccounts173 at jgoettgens.de> wrote: > > >> I'm not able to install asterisk whenever I hit make command I get below >> error: >> >> make[1]: *** No rule to make target `../main/modules.link', needed by >> `asterisk'. Stop. >> make: *** [main] Error 2 >> >> > Just

...Error 2 when compile asterisk To get around this, just delete following line in file makeopts.embed_rules EMBED_LDSCRIPTS+=../main/modules.link Source: http://showmyroutes.com/wordpress/?p=500 Sincerely, M azhar http://www.nicacresults.com On Mon, Apr 13, 2015 at 1:10 PM, akhilesh chand <omakhileshchand at gmail.com> wrote: > Hi folks, > > > I'm not able to install asterisk whenever I hit make command I get below > error: > > make[1]: *** No rule to make target `../main/modules.link', needed by > `asterisk'. Stop. > make: *** [main] Error 2 > > &gt...

Hi Folks, I'm trying to register softphone(3CX Phone) in AWS Cloud but I'm not able to register I got below screen. [image: Inline image 1] Register Screen for 3CX Phone [image: Inline image 1] Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Check if asterisk is running or not first . If asterisk is running check iptables ( firewall ) might be blocking the connection . You can see listening ports with netstat -upln command On Mon, Apr 20, 2015 at 10:01 AM, akhilesh chand <omakhileshchand at gmail.com> wrote: > Hi Folks, > > I'm trying to register softphone(X-lite) but I'm not able to register > softphone whenever I'm trying to register softphone I got below error > > [image: Inline image 1] > > Is there any document/guide line where I will ge...

On Mon, Apr 20, 2015 at 1:58 AM, akhilesh chand <omakhileshchand at gmail.com> wrote: > Chain INPUT (policy ACCEPT) > target prot opt source destination > ACCEPT all -- 0.0.0.0/0 0.0.0.0/0 state > RELATED,ESTABLISHED > ACCEPT icmp -- 0.0.0.0/0 0.0.0.0/0 > ACCEPT all -- 0.0....

Dear All, I have six different campaign and 5 different agent have login on that campaign.*Same thing i have done using agi and database,i never use queue management on this scenario. Agent** can also shuffling one campaign to anther campaign. * Now i want to do some work with queue.I want to use single queue to managing this. Eg: campaign Agent Login A a_1,a_3

Dear all, I have two system Sys A and Sys X. Sys A is normal PC. Sys X have installed asterisk 1.6 and i want register(or reserved) sip extension(like 4001,4002,4003..) through Sys A(Sys A have some ip address) but i don't use any soft-phone means i want to write Perl or php(any language) script to register sip extension. Suppose to 4001 is reserved with Sys A. 4002 is reserved with

Hi folk, I'm trying to register an extension through softphone and got stuck.I got below error:- [Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via: missing sent-by in Via header [Nov 22 07:31:28] ERROR[3522]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("", "(null)", ...): Name or service not known [Nov 22 07:31:28] WARNING[3522]: chan_sip.c:16056

Dear All, I have pri with E1 facility that have 30 line and 100 pri number which is provided by service provider.Number started like 23568561,23568562,23568563 and so on. Service provider provide last four digit number for did mapping like 4561,4562,4563. exten => 8561,1,Dial(SIP/4001 at 192.168.14.110,120,tT) exten => 8561,n,hangup() exten => 8562,1,Dial(SIP/4001 at

Dear All, I want to disable internal call facility.Means agent(4002) does not make call to agent(4003) or other extensions. Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131016/e1c5734f/attachment.html>

What is the easiest way? And how can it be implemented? I thought to something like: 1. I request a page to the webserver 2. Perl sends to asterisk a number to dial (Perl and asterisk are running in the same machine) 3. Asterisk calls the phone or 1. A Perl sip client registers to remote asterisk server 2. Perl sip client sends to asterisk the number to dial 3. Phone rings

Akhilesh, I have implemented several ec2 instances with both sip and iax2 and have no problems with xlite or hard phones. Have you already opened the ports in the vpc security group on the Amazon side? Let me know is I can help. --James On Apr 24, 2015 3:34 AM, "akhilesh chand" <omakhileshchand at gmail.com> wrote: > Hi Thomas, > > Could you tell how can I change the protocol of corresponding port means > 5060 is configured with tcp protocol I want to configured with udp. When I > execute nmap -p5060 xx.xx.xx.xx I got below output > > > [root at ip-172-31-32-11...