similar to: Follow Me CID

Hi all I installed DAHDI Version - 2.6.1 DAHDI Tools Version - 2.6.1 libss7-trunk Asterisk 11.0.1 from source on Fedora 12 x86_64. Now i`m unable to load chan_dahdi and libss7: myserver*CLI> module load chan_dahdi.so ?ERROR[10124]: chan_dahdi.c:17842 process_dahdi: Unknown signalling method 'ss7' at line 37. myserver*CLI> module load libss7.so Unable to load module libss7.so

Can someone help me with upgrading to the lastest version. I am using the same sip.conf file, but the headers have changed and registration fails. Has something change in the conf file that would cause this? Notice 1.2.5 has no Authoization at all... Regards, Jason Version 1.0.9 --------------------------- REGISTER sip:voip-ca35323.ocn.ne.jp SIP/2.0 Via: SIP/2.0/UDP

The Asterisk Development Team has announced the release of libss7 2.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libss7 The release of libss7 2.0.0 resolves several issues reported by the community and would not have been possible without your participation. Please note that this version of libss7 has been released in anticipation of what

The Asterisk Development Team has announced the release of libss7 2.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libss7 The release of libss7 2.0.0 resolves several issues reported by the community and would not have been possible without your participation. Please note that this version of libss7 has been released in anticipation of what

Hello All. I am stumped, please help me out.. I have the following setup: VOIP provider = VOIP GW (asterisk GW1) = VOIP server (asterisk - VS1) The gateway is there to get around the limitations running on the VOIP server. I can call out from and receive calls VS1 no problems at all. However, when I try and redirect an inbound call out via the GW, it drops out. I have found that if I

Hi, well, swift response! Qualcomm has a patched qpopper (2.5) Greetings, Jan-Philip Velders <jpv@jvelders.tn.tudelft.nl> ---------- Forwarded message ---------- Date: Mon, 29 Jun 1998 21:43:18 -0700 From: Praveen Yaramada <pyaramad@QUALCOMM.COM> To: BUGTRAQ@NETSPACE.ORG Subject: Patched Qpopper2.5 release Notification. Hello Folks, As you are already aware that qpopper

Hi Are there any known security holes or necessary precautions in using port forwarding with ipportfw? I'm planning on forwarding ports from an outer firewall/router (connected to the Internet) to a host in the DMZ, then on from the DMZ host to the inner firewall, and finally from the inner firewall to some host on the inside. Thanks, Jens jph@strengur.is From mail@mail.redhat.com Wed

I am wondering if anyone knows of a way to do this, as it would be much more meaningful for our CDR reports. We use FreePBX under the Elastix distro. We are able to set the CALLER's CID on inbound calls by using the "Asterisk Phonebook" module in FreePBX, then configure the Inbound Route settings to use it for CID. I haven't seen anything like this to apply those same rules to

The Asterisk development team has released new versions of three libraries used with Asterisk. They are: libpri-1.2.8: This release contains a number of bugfixes that had been unreleased for months, along with clarification of the licensing of the source code. The change log is here: http://downloads.digium.com/pub/telephony/libpri/ChangeLog-1.2.8 libpri-1.4.7: This release contains primarily

Hei! I'm trying to send special characters out to ss7 link, but libss7 seems to convert them to zeroes. The challenge is that our service provider demands some of the regional numbers to be sent in format D0+number. When I use D in front of the number in dialplan, libss7 replaces it with 00, So I have a dial string: exten => _[A-Z].,1,Dial(DAHDI/g1/DD0501,,g) But in SS7 trace I

Hi all! I have trouble with connection to AMI 1.1 wich enabled on Elastix "*Asterisk Call Manager/1.1* *Action: Login Username: admin Secret: qweasd123* *Response: Error* *Message: Missing action in request*" Elastix versions: "* Kernel* * Linux(x86_64)-2.6.18-348.1.1.el5* * Elastix* * elastix-2.4.0-1* * elastix-portknock-0.0.1-0* * elastix-agenda-2.4.0-1* *

Hi. Dont know if this is an Asterisk or Dahdi or LibSS7 Error. So Im writing to Asterisk List. If somebody knows where to search (dahdi lists or libSS7 lists) will be appreciated. Im getting this error after a certain time, My config is: Hardware: 3 Digium Quad E1 TE4XXP libss7 version: SVN-branch-1.0-r286 DAHDI Version: 2.4.0 Echo Canceller: Asterisk 1.6.2.14 CentOS release 5.5 (Final) Kernel

We have a problem in that calls are dropped after 15 minutes (on both internal and out going calls, incoming calls do not seem to have that limit) How do we fix it? This is the version on that PBX Kernel Linux(x86_64)-2.6.18-371.1.2.el5 Elastix elastix-2.4.0-8 elastix-a2billing-1.9.4-5 elastix-addons-2.4.0-10 elastix-agenda-2.4.0-14 elastix-asterisk-sounds-1.2.3-1

Samba share access is not working for trusted domain users. I have updated krb5.conf with both domains realm and in smb.conf trusted domains default value not changed. But still share access is not working for trusted domain. # less lnxbrisglog01.02188394-667794/installed-rpms | grep samba samba-3.0.33-3.40.el5_10-x86_64 Tue 29 Sep 2015 03:25:42 PM AEST samba-common-3.0.33-3.40.el5_10-x86_64 Tue

This Wednesday, November 26th, the Toronto Asterisk Users Group invites all in the area to join us for a telephony workshop and talk sponsored by Sangoma Inc.[1] Jose Landivar, co-founder of PaloSanto Solutions[2], creators of Elastix, will be running a "getting started" workshop on Elastix, followed by a talk discussing how it differs from other Asterisk-based distributions, and a

Hi, I?m trying to use SS/ in Asterisk. I'm thinking in chan_ss7 and libss7, and I want to know some other experience with this. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100121/f8c4937e/attachment.htm

Dear community, Please help. I've been looking around the internet (and in this great forum) for help with DUNDi setup between servers (I'm using Elastix) and while I can get my servers to lookup extensions on each other very well, I have not been able to successfully make calls between servers. For my test environment, I have 3 servers setup for now, and these are the steps I've

Hello List, To whom it might concern: I have been working in some SlackBuilds (script for making Slackware Packages) for my personal use, but thought they might be useful for someone else here. Beside of the exceptional distributions used so far (CentOS, Debian, Ubuntu, etc.), you might want to test Asterisk on a Slackware Linux box, as it offers outstanding stability and flexibility as

---------- Forwarded message ---------- From: Juan Carlos Huerta <juancarloshuerta at gmail.com> Date: 07-abr-2009 13:41 Subject: Re: [asterisk-r2] MFCR2 With Unicall - can unicall affect zaptel commands? To: asterisk-r2 at lists.digium.com Please wirte to asterisk-users at lists.digium.com to get help about this problem. Juan Carlos ~ Lo que no te mata te fortalece ~ On Tue, Apr 7,

Hi; I used vicidial for call center and I would like to try elastix. Can someone advise about the advantages? Does Elastix has a screen for the agent to login/logout from their PC and deal with the inbound/outbound calls and Integrated with the CRM? Regards Bilal