search for: libss7

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 lo...

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...

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...

...ChangeLog-1.2.8 libpri-1.4.7: This release contains primarily only clarification of the licensing of the source code and some minor build system fixes. There is no need for users of version 1.4.6 to upgrade. The change log is here: http://downloads.digium.com/pub/telephony/libpri/ChangeLog-1.4.7 libss7-1.0.1: This release contains a number of bugfixes, along with clarification of the licensing of the source code. The change log is here: http://downloads.digium.com/pub/telephony/libss7/ChangeLog-1.0.1 Thanks for using Asterisk! -- Kevin P. Fleming Director of Software Technologies Digium, Inc...

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)...

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

Hi Team, i have been facing issues with sangoma card with 16 E1. used LibSS7 asterisk 1.6 with 8 E1 the links are stable , but moment i add another card of 8 E1 for to support 16 E1. link keeps fluctuating any idea why ? Please help Thanks Vinod Dharashive -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/piperma...

In reference to an old post from 2002: http://www.marko.net/asterisk/archives/0203/0103.html How does one go about doing this? Also, what is the present status of the OpenSS7 stack in Asterisk? What can it do now? And is there any possibility in the future of developing a DS3 card for it, if only for the purpose of mostly DACSing? Which is still a level of intelligent call control on the

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: Asteri...

I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 with a TE120P card. I can't make any connection over the T1. From CLI: ERROR[26017]: chan_dahdi.c:14300 process_dahdi: Unknown signalling method 'pri_cpe' at line 37. cat chan_dahdi.conf cat chan_dahdi.conf [trunkgroups] [channels] language=en ;internationalprefix = 00 ;nationalprefix = 0 context=from-pstn switchtype=national

...sed to transport ss7 signaling. - FXS interface in PBX2 -----------------> connected to -----------------> FXO interface in PBX1 =============> used to transport voice between the two PBXs. This connection can be replaced by a simple SIP trunk. Is this scenario possible with libss7 and asterisk. If yes, please give some instructions and tips. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100416/276cb25e/attachment.htm

...stability and flexibility as well. I. The scripts are located at: http://packages.eslackware.com/slackbuilds/asterisk/ II. So far, they consist of the following: (Listed in recommended building order) a. newt-SlackBuild.tar.gz NewT Libraries b. libpri-SlackBuild.tar.gz LibPRI c. libss7-SlackBuild.tar.gz LibSS7 d. dahdi_linux-SlackBuild.tar.gz DAHDI Linux e. dahdi_tools-SlackBuild.tar.gz DAHDI Tools f. unixODBC-SlackBuild.tar.gz UnixODBC (entire ODBC API, Drivers, and tools) g. mysql_connector_odbc-SlackBuild.tar.gz Connector/ODBC (standardized database dri...

Hi, I have setup 2 asterisk talking? a single mysql cluster. I'm also using realtime db. I've setup sip peering between the two asterisk servers. [asterisk-1] insecure=port,invite type=peer host=201.202.203.204 context=from-asterisk-1 [asterisk-2] insecure=port,invite type=peer host=201.202.203.205 context=from-asterisk-2 scenario: ext 100 registers on Asterisk 1 ext 200

Dear all, i wan to configure digium te412p card as ss7 interface. i installed libss7,dahdi and asterisk. Do i need to install lip pri also to work on ss7? best regards. Kashif -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090418/7fa92295/attachment.htm

Several regulars from the VUC will be there, some of us are arriving Tuesday night. Anyone else considering the trip? Post here or contact me off list so we can meet. /r

Hi, I am trying to set up asterisk with ss7. Whenever I try to load module chan_dahdi.so, I get the error [Mar 26 17:33:27] ERROR[10437]: chan_dahdi.c:10458 mkintf: Unable to find linkset -1 I have compiled dahdi, libss7, asterisks (am using asterisk 1.6) in that order. Have already set signalling to ss7 in dahdi_channels.conf How do I sort this out? Thanks for your help in advance. Peter. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/aste...

...numbers drops after few sec. where as, with same configuration , and making obd on to pri it works properly. Also i had words with Idea team, they says release is from my point code, whereas on debugging at my asterisk server i received release from their point code. using :- dahdi-tools-2.3 , libss7-1.0 , asterisk-1.6.2. Any suggestions ...?

...have recently implemented asterisk as the office PBX, and as a GUI, we chose Elastix. All seems quite easy and am very pleased with the way Asterisk is running and handling calls. To summarize briefly, our setup looks like this: LAN -> Asterisk PBX (Elastix) -> Asterisk TDM (basic CLI with libSS7) We get a whole 3 digit number series from our telco. So, like, +1 312 765 7xxx (just an example), where we get to choose XXX on our own. To simplify things, we have forwarded the full incoming extension to "Asterisk PBX", and have assigned full numbered SIP IDs to our users. And so f...

Hello We are experiencing some difficulties with T38 faxing. I have a Asterisk 11.5.0 with libss7 and Sangoma A104DE digital interface card . The operating system is Centos 6 We are using this server to terminate calls to Telco. So calls are coming to asterisk from sip and we are sending calls to Telco with Dahdi. (It is a one way interconnection only from asterisk to telco ,not from telco to...

...rce-1.6.0.6-82.2.x86_64.rpm /srv/distro/repo/network:/telephony:/asterisk/openSUSE_11.1/x86_64/libpri-debugsource-1.4.7-9.2.x86_64.rpm /srv/distro/repo/network:/telephony:/asterisk/openSUSE_11.1/x86_64/flite-devel-1.3-8.2.x86_64.rpm /srv/distro/repo/network:/telephony:/asterisk/openSUSE_11.1/x86_64/libss7-devel-svn-31.2.x86_64.rpm /srv/distro/repo/network:/telephony:/asterisk/openSUSE_11.1/x86_64/zaptel-debugsource-1.4.12.1-32.2.x86_64.rpm /srv/distro/repo/network:/telephony:/asterisk/openSUSE_11.1/x86_64/iksemel-devel-1.3-8.2.x86_64.rpm /srv/distro/repo/network:/telephony:/asterisk/openSUSE_11.1/x8...