similar to: Read Telnet Packet

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 James, Please let me know how could I implement for sip.I will appreciate your help. Regards Akhilesh On Mon, Apr 27, 2015 at 7:01 PM, James Cass <jcass78 at gmail.com> wrote: > 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

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-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC Nmap scan report for

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

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 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

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

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>

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

Hi Guenther, When I executed nmap -p5060 <xx.xx.xx.xx> I got below output. [root at ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC Nmap scan report for ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com (xx.xx.xx.xx) Host is up (0.00080s latency). PORT STATE SERVICE 5060/tcp filtered sip Nmap done: 1 IP address (1 host

Dear All, I'm installing the asterisk-1.6.2.24 in Centos 5.3, whenever i'm running following command ./configure I got below error: configure: *** XML documentation will not be available because the 'libxml2' development package is missing. configure: *** Please run the 'configure' script with the '--disable-xmldoc' parameter option configure: *** or install the

Hi Helvio, Could you tell me what is process to setup an environment for IAX. Regards Akhilesh On Fri, Apr 24, 2015 at 4:25 PM, Helvio Junior <helvio.listas at gmail.com> wrote: > Hi Akhilesh, > > SIP protocol use port 5060 (default) and many other ports to stablish > calls. You need to check if there is AWS firewall rule that allow your > communication from your client

Hi Guenther, Thanks for ur reply I have concern from long time I'm not able to login through softphone with AWS Cloud.Please let me know is there any document or guide line for the same. Regards Akhilesh On Fri, Apr 24, 2015 at 1:26 PM, Guenther Boelter <gboelter at gmail.com> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 04/24/2015 03:34 PM, akhilesh

Hi Guenther, What did you recommend to me, I did accordingly but there is no log showing in asterisk CLI. I'm getting same problem. Regards Akhilesh On Mon, Apr 20, 2015 at 6:05 PM, Guenther Boelter <gboelter at gmail.com> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 04/20/2015 12:31 PM, akhilesh chand wrote: > > Hi Folks, > > > >

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

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

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:

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

Hi, Try as a first step a tcpdump capture to verify if the softphone is actually sending the register message to the server. For me it seems like the softphone is not able to reach the server ! Best Regards, On Fri, Apr 24, 2015 at 10:55 AM, Helvio Junior <helvio.listas at gmail.com> wrote: > Hi Akhilesh, > > SIP protocol use port 5060 (default) and many other ports to stablish