similar to: Register Sip extension with out Sip phone

I'm kind of at a loss to diagnose problems like this, yet we get them a lot. - The ATA (Thomson 784 in this particular case) is logged into the Asterisk server. 'sip show peer' shows their IP address, port, and useragent. - The ATA is connected directly to the internet (no NAT, but the sip configuration has nat=always) and logs in to our server, which is also directly connected to the

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:

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>

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, I am working on developing a professional Marketing Materials for my systems. I plan on using a very good(expensive) company to do that so splitting the costs with several people would be nice. Let me know if you are interested on taking part in it. robert -------------- 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

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 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, 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 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

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 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

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

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, is there anything going with VoiceXML in asterisk??? is this the list to query regarding this or should I put this on the dev list? thanks, dave cantera

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

Hi Greeting to all you out there. I am new at asterisk, I have been working with PLMN platforms telecommunication for 5 years with NSN and Huawei. We have recently built an asterisk PBX with Trixbox and connected it to our MSS using Digium E1 cards(ISDN). Everything went smoothly as there are tons of information out there, except for the TON number. If you have worked in Telecommunication you

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