similar to: Grandtream Conference issue

Displaying 20 results from an estimated 1000 matches similar to: "Grandtream Conference issue"

2008 Jan 21
1
asterisk-addons-1.6.0-beta1---Error
Hi, I'm trying to install asterisk-addons-1.6.0-beta1 on my machine. But getting following error during make: [root at Cancer asterisk-addons-1.6.0-beta1]# make make[1]: Entering directory `/usr/src/asterisk/asterisk-addons-1.6.0-beta1' [CC] app_addon_sql_mysql.c -> app_addon_sql_mysql.o app_addon_sql_mysql.c: In function `aMYSQL_connect': app_addon_sql_mysql.c:266: error:
2007 Jul 17
2
media not accpetable with outgoing call on cisco
Hello, I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec in my ata the the GW return a media not acceptable error. but If i add the g729 codec the all is ok. I see in the config of the cisco where to define codec for imcoming call but not for outgoing *Jul 17 15:57:02.604: Received: INVITE sip:0041787518551 at 192.168.0.110 SIP/2.0 Via: SIP/2.0/UDP
2008 Jan 20
6
IAX softphone
Hi All; I tried Firefly softphone with IAX and it gave very poor quality. Any one advise a good strong softphone that can work with IAX fine? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
2005 Feb 22
3
Call Manager Express Peer
I have the following configuration and am obviously missing something small that is causing * not to work as expected. I have the following defined in sip.conf [ccme-in] type=peer host=10.0.9.1 context=devel_in disallow=all allow=alaw nat=no canreinvite=yes qualify=yes and [devel_in] is defined in extentions.conf However when I try to call via the dial peer I have configured on the cisco
2011 Mar 20
6
PATCH: Hugepage support for Domains booting with 4KB pages
We have implemented hugepage support for guests in following manner In our implementation we added a parameter hugepage_num which is specified in the config file of the DomU. It is the number of hugepages that the guest is guaranteed to receive whenever the kernel asks for hugepage by using its boot time parameter or reserving after booting (eg. Using echo XX > /proc/sys/vm/nr_hugepages).
2012 Dec 09
3
Unable to boot Linux kernel in from Syslinux 6.00-pre efi64
Hi, Line 762 of efi/main.c (git firmware branch) should be changed to if (hdr->version < 0x20b) { Without this change syslinux efi64 gives out "handover protocol unimplemented error" and fails to boot. With this change syslinux boots the kernels fine (tested with 3.6.9 and 3.7rc6 kernels). Please fix this. Thanks in advance. Regards. Keshav
2007 Jul 19
2
Upgrade Procedure
Hello All, I would like to upgrade my recently installed Asterisk 1.2.21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh
2011 Jan 10
9
Hugepage Support
hi, I tried to make huge page request in Fedora x86_64 PV guest using xen 4.1 unstable and it crashed(crash info given below) I had enabled superpages in config file I had also set hugepages parameter at boot time for the PV Dom U By excuting # cat /proc/mem_info | grep Huge gave me that there are 10 free huge pages available , still the domain crashed. [ 86.403654] BUG: unable to handle
2007 Jul 18
3
how to use call transfer
Dear all I have beginer in Voip and i have configured Asterisk server with 100 IP SIP phone ( SNOM ) everything is fine but problem is how to transfer call from one user to other means i call to some one and then someone want to transfer call to another person how it is possible i have also try with feartue.conf but it is now working i have also read document on voip-info website
2011 Apr 16
3
any experience with cisco media gw with fax???
Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I already have t38pt_udptl=yes in sip.conf Thank you.
2011 Aug 22
1
[PATCH] protocol.h: Remove unused locate_protocol()
From: Matt Fleming <matt.fleming at intel.com> locate_protocol() has never been used by efilinux and results in the following build error when compiled with gnu-efi <= 3.0i, In file included from entry.c:38:0: protocol.h: In function 'locate_protocol': protocol.h:62:31: error: 'EFI_BOOT_SERVICES' has no member named 'LocateProtocol' Reported-by: KESHAV P.R.
2010 Dec 30
2
memdisk + grub2 consumes lot of memory
Hi all, I think I have asked about this before but I do not know whats happening. I have problems booting syslinux in my GPT disk which I have asked in a separate topic "Boot Error GPT partition" . I loaded https://github.com/skodabenz/Tianocore_DUET_memdisk_compiled (append options - floppy ro nopass) . Using syslinux when it was booting properly, did not lead to loss of RAM.
2010 Dec 30
3
Boot Error GPT partition
Hi all, I was using grub2 for booting Archlinux x86_64 in my GPT Internal HDD (/dev/sda) then I switched to syslinux/extlinux. It was working fine but suddenly it staring showing "Boot Error" message on screen. I googled for a solution and tried all syslinux versions from 4.04-pre4 down to 4.03-stable but non of them worked. I don't know what suddenly changed caused this
2013 Oct 29
1
syslinux/isolinux 6.2 "Error: Couldn't read the first disk sector"
On 29 October 2013 11:39, Steven Shiau <steven at nchc.org.tw> wrote: > My bad, I meant version 6.02, not 6.2 > > Steven. > > On 10/29/2013 11:28 AM, Steven Shiau wrote: > > After switching to version 6.2, I am having a problem to use chain.c32 > > to boot a local disk. > > My settings: > > ====================== > > label local > > MENU
2004 Apr 30
3
Asterisk <--> Cisco router
What codec should be used to connect a * box to a cisco router which has a t1 with 24 trunks coming in? My router voip dial plan looks like this: dial-peer voice 2 voip destination-pattern [1,2,,3,5,8].. session protocol sipv2 session target ipv4:10.x.x.x dtmf-relay cisco-rtp codec g711ulaw no vad ! The problem I have is when more than one call is on it, sometimes the quality gets very
2003 Sep 24
6
Cisco 2600 and ASTERISK
Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Jul 18
1
Issue in insatlling addons-1.4.2
Hi, I'm using Asterisk-1.4.7.1. Everything was working fine. Now I'm trying to Install Asterisk-addons-1.4.2. The procedure I followed is as... # cd asterisk-addons-1.4.2 #./configure #make menuselect #make #make install Everything is going fine except make install. I've tried many times, but the same error I'm gettiing--- The error is--- asterisk-addons-1.4.2]# make install
2005 Oct 13
2
Sample cisco config for cisco 7206
I see a lot of comments but no actual show runs. Can someone post a 7206 config. I am having a dickens of a time getting calls to pass. I currently have the following loaded. Cisco IOS Software, 7200 Software (C7200-IK9O3S-M), Version 12.3(8)T6, RELEASE SOFTWARE (fc2) Thanks !!! Jerry -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 /
2005 Jul 20
1
Re: help me in printer driver
You mean u have no idea to develop printer driver keshu >From: David Collier-Brown <David.Collier-Brown@Sun.COM> >Reply-To: samba@lists.samba.org >To: keshav chaudhary <keshu82_sh@hotmail.com> >Subject: Re: help me in printer driver >Date: Wed, 20 Jul 2005 09:44:53 -0400 > >[Sent to samba-technical by mistake] > >No, but I've forwarded this to the
2005 Jun 30
1
Cisco Voip Question
Does anyone in here know how to setup auto negotiation between g729 and g711ulaw on a cisco 5400? I would imagine it would be the same on a 3660. The problem I am having is natively the call is setup for g729 however when the call is transferred to voicemail it uses ULAW so when the cisco tries to connect to the voice mail I get a SIP error that the codec couldn't be negotiated. I need