search for: arminco

Hi all, I want to install clang on CentOS 7. After executing "yum install clang", it outputs: [root at hp ~]# yum install clang Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: centos.mia.host-engine.com * elrepo: elrepo.mirrors.arminco.com * extras: centos.arvixe.com * updates: centos.eecs.wsu.edu No package clang available. Error: Nothing to do Doesn't CentOS 7 support rpm install clang? Thanks in advance! Best Regards Nan Xiao

...error in google, but the suggestions given there is not working for me. The same message showing for both software pakage. root at localhost tmp]# yum install grace Loaded plugins: fastestmirror, langpacks Loading mirror speeds from cached hostfile * base: ftp.iitm.ac.in * elrepo: elrepo.mirrors.arminco.com * epel: epel.mirror.net.in * extras: ftp.iitm.ac.in * rpmforge: apt.sw.be * updates: ftp.iitm.ac.in No package grace available. Error: Nothing to do root at localhost tmp]# Please suggest me to how to resolve the problem ? Thank you in advance for help Regards Rahman '''...

...want to install clang on CentOS 7. After executing "yum install > clang", it outputs: > > [root at hp ~]# yum install clang > Loaded plugins: fastestmirror > Loading mirror speeds from cached hostfile > * base: centos.mia.host-engine.com > * elrepo: elrepo.mirrors.arminco.com > * extras: centos.arvixe.com > * updates: centos.eecs.wsu.edu > No package clang available. > Error: Nothing to do > > Doesn't CentOS 7 support rpm install clang? Thanks in advance! > > Best Regards > Nan Xiao > ___________________________________________...

Greetings, Is there a way to tie a specific sip username to a IP address when authenticating against mysql sipfriends table? (USE_MYSQL_FRIENDS=1 USE_SIP_MYSQL_FRIENDS=1 in channels/Makefile) The reason is that I'm using Wellgate FXSes that have second/third/fourth FXS ports bugged when I use a password, but work ok when there is no password. Linking the username to a specific ip could

I'm authenticating against sipfriends in MySQL, and have just noticed that none of the below commands return any output: sip show users sip show inuse sip show active sip show subscriptions Is this a bug or something wrong on my side? I'm using the stable 1.0 cvs Vahan

Anyone else has problems with CVS HEAD's from today with voicemail hanging up silently without any debug/error messages when checked? It also keeps insisting that the user's voice mailbox is full and can't store more messages even if I clear/rebuild the /var/spool/asterisk/voicemail stuff. I've tried falling back to voicemail.conf entries from realtime voicemail with the same

Has anyone used this board as an Asterisk server? http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=H&IPMI=Y I'm mostly interested about the possible compatibility issues this board may have with the AEX800 card.

Hi, I was wondering if anyone has successfully configured two lsi/3ware 9750-4i series controllers for multipathing under CentOS 5.7 x86_64? I've tried some basic setups with both multibus and failover settings, and had repeatable filesystem corruption over a iscsi(tgtd) or nfs3 connection. Any ideas? Vahan

...rom: <sip:010527911@195.250.77.70>;tag=d8424c26a4 So here it is: it uses the detected caller-id instead of the FXO's username. Still, later on the auth challenge step it resends the invite with the proper auth info: Proxy-Authorization: Digest username="582760", realm="sip.arminco.com", nonce="1886728b", uri="sip:411@195.250.77.70", response="487250bb2f1f17a8b15e9ad727e87a6f", algorithm=MD5 ..asterisk rejects the call with Failed auth on 010527911@195.250.77.70 :( Is there a limitation in Asterisk and it uses the "From" address...

Greetings, I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. So far everything is working as expected, except the fact only ulaw and alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both

I've setup the voicemail that auths against the mysql db. Now, everything works ok, except voicemail() calls fail with Sep 26 18:09:34 WARNING[157070336]: app_voicemail.c:1517 leave_voicemail: No entry in voicemail config file for '' all my users are in 'sip' voicemail context, but adding context to it: voicemail(@sip) doesn't help.. while if I put a vmbox # to it, it

I'm playing with codecs at the moment and have found some notices errors when x-lite/pro connects to asterisk with Speex or ILBC. Initially I was getting garbled sound, but after changing magic number for both codecs to 97 (as per http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite and http://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to get normal voice. BUT,

I've just made (on FreeBSD 4.10) cvs co asterisk gmake gmake install asterisk -vvvvvvc and got a new error message (was working ok before the cvs) [codec_gsm.so]Oct 6 17:23:16 WARNING[135192576]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/codec_gsm.so: Undefined symbol "Short_term_analysis_filteringx" Oct 6 17:23:16 WARNING[135192576]: loader.c:429 load_modules:

Greetings, For the past 2 months I've been struggling with registration problems with asterisk+external FXS/FXO gateways (www.addpac.com) that use RFC3665 re-registration procedure. This problem occured for devices with more than one FXS port with a set non-empty password. Those gateway attempt to re-register after the initial register timeout period expires fully compliant with

http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150&tid=147&tid=136 http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/ -------------- next part -------------- A non-text attachment was scrubbed... Name: vahan.vcf Type: text/x-vcard Size: 287 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/357c6cce/vahan.vcf

Greetings, Since the past week I've started receiving the following warnings on my asterisk servers (FreeBSD / CVS-HEAD). This warning manifests itself with x-lite/x-pro/eyebeam clients as well as sipura devices. All of them have qualify=yes in their settings. Jul 18 22:52:01 WARNING[73576]: chan_sip.c:939 __sip_xmit: sip_xmit of 0x8a3401c (len 483) to 195.x.y.28 returned -1: Address

What is the proper way of adding hints to multiple extensions? In my case extensions are the same as the sip usernames, while as per http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence exten => 1234,hint,SIP/1234 works, exten => _1XXXX,hint,SIP/${EXTEN} doesn't. Not sure if I can even use ${EXTEN} here... Any hints? Vahan -------------- next part -------------- A

Hi, I'm looking for a full list of xml provisioning variables of the SPA-2100/3000. Currently the Sipura website has example XMLs only for the SPA-841 [1] and SPA-941 [2]. I'm mostly interested in the CallerID type selector variables and whatever variables control the PSTN<->VoIP settings. Sipura Configuration website form field names are numeral only. :( [1]

Hi, We are processing lots of calls and I want to filter these that have incomplete numbers sent with a proper SIP response. These numbers are not in the local dialplan by themselves, so I'm trying to find a way to generate 484 "Address Incomplete" SIP response based on the length of the extension called. Congestion response is too lossy of the original cause and doesn't

I'm loosing hair at cosmic speed now for the past 10 days. Welltech's Wellgate 38xx/35xx FXO/FXS SIP hardware versions seem to have very buggy firmware possibly due to hastely done porting from H.323 firmware. Is there anyone on this mailing list who was able to: 1. setup a 35xxA FXS with all ports authenticating properly with *? or 2. setup a 38xx FXO to work as dial-in from pstn to