similar to: G.723.1 codec

This *should* be something that's easy to fix, but apparently I'm not doing something right. Our SIP long distance provider is telling us to only use formats G.723 and G.729, so I've set up their trunk configuration in sip.conf as such: [t564] type=friend host=XXX.XX.56.4 context=default disallow=all allow=g723 allow=g729 However, the Dial application gives the following error:

Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI> core show version Asterisk

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the

Hi, Companies that offer telephone service to the public are obliged to offer tapping to all kind of authorities. Does anyone know how to tap in Asterisk? I.e. record (or copy) a conversation based upon their telephone number? Thanks Dan _________________________________________________________________ Stay in touch with absent friends - get MSN Messenger http://www.msn.co.uk/messenger

With 2.6.31-rc5 in a KVM guest using dm and virtio_blk, we see the following errors: end_request: I/O error, dev vda, sector 0 end_request: I/O error, dev vda, sector 0 The errors go away if dm stops submitting empty barriers, by reverting: commit 52b1fd5a27c625c78373e024bf570af3c9d44a79 Author: Mikulas Patocka <mpatocka at redhat.com> dm: send empty barriers to targets in

With 2.6.31-rc5 in a KVM guest using dm and virtio_blk, we see the following errors: end_request: I/O error, dev vda, sector 0 end_request: I/O error, dev vda, sector 0 The errors go away if dm stops submitting empty barriers, by reverting: commit 52b1fd5a27c625c78373e024bf570af3c9d44a79 Author: Mikulas Patocka <mpatocka at redhat.com> dm: send empty barriers to targets in

Hello, I am relatively new to Asterisk and I need to compile the G.723.1 codec for Asterisk. I downloaded the ITU source code, placed it in the codecs directory, but apparently Asterisk needs a rather different library than the one provided from ITU. As I've seen in the mailing list archives, there are quite a few users who were able to compile G.723.1 in *, so, could someone kindly share it

Ok, After following BJ's advice and removing ogg.so I then got a pbx_realtime.so error in the same fashion. I removed that file, and then the next and then the next as you can see in the log below. I think something is not right. duh here is my sign..lol...but I am not sure even where this ast_register_file_version flag is in a config file or what step I have missed. I am doing a VOIP only

Hello guys Does anyone use lvm2 backends for domU storages ? I do and I wanted to use lvm''s snapshot feature to make backups of domUs in background but I got the following problem. When I create a snapshot LV and copy data from it to backup storage it works perfect. Then I do umount and then lvremove. lvremove asks me if I really want to remove the volume and then just hangs forever.

Can I configure Samba Wins (ver. 2.2.2), installed on Sun Solaris 5.8, as a partner push-pull of Microsoft Windows WINS service installed on Windows 2000 server ? Thanks Francesco romano -------------- next part -------------- HTML attachment scrubbed and removed

Hello Michael, Here is the BackTrace of the program which i forgot to attach BACKTRACE OF Asterisk -vvc #0 0x42074d60 in _int_realloc () from /lib/tls/libc.so.6 #1 0x420738c4 in realloc () from /lib/tls/libc.so.6 #2 0x47c7da89 in PAbstractArray::SetSize(int) () from /home/sip/pwlib/lib/libpt_linux_x86_r.so.1.5 #3 0x47c7cf4d in PContainer::SetMinSize(int) () from

I added the line "allow G723.1" in my sip.conf general config, and from a bridge connection which gives silence, I have progressed to the error message below, and the call gets rejected. help!! Dave ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant Expressa 723@216.52.153.207 : Go2Call SIP gateway -- Executing

Hello Have a trouble. I have mashine with two interfaces. Samba running at both with the same name of computer. In wins-servers wins.dat i see next string: "GRAND#20" 1044171787 192.168.1.1 192.168.2.1 46R It's ok. But for ALL mashines in any interfaces with any IP (192.168.1.X or 192.168.2.X) returned first IP in list: ------------------ [2003/01/30 11:03:21, 3]

Hi Stephen, Thanks very much for getting back to me. My problem is described below. Any help would be greatly appreciated. Thanks, James Hi, Sorry for bothering you because I know that your time is voluntary, but I would really appreciate some help. I work in a hospital in part of Ireland?s national health service, a service which was struck by a massive cyber attack a couple of years ago. Since

----- Original Message ----- From: "Jess Magnaye" <jess@arretni.com> To: <wipe_out@users.sourceforge.net> Sent: Tuesday, January 06, 2004 3:19 PM Subject: Re: [Asterisk-Users] Pls confirm > Is the format "allow=g723.1" in sip.conf valid? > > somehow i cannot get it working to do g723 passthru. also, i've read that > doing g723 will disable

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

Please, help I could not compile g723 format with pwlib-v1_4_11 and openh323-v1_11_7 I'am planning to use h323 channel driver, because of it that versions of libraries have been cvs'ed from openh323.org I am getting next compile errors: gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6

Folks! I would appreciate if someone could send me a simple working h323 configuration file oh323.conf that is part of asterisk@home installation. I have tried to use the oh323.conf content listed on WIKI but it is just not working as my H323 endpoint ( PA168 based ATCOM Phone) cannot register. I need a working example of this file for similar phone. Seshu

Hi all, i have big problems to make a h323 call over the gatekeeper from my provider. The provider demanded following account data: H323 ID: XXX-XXX-XX-X DetinationNumer: XXXXXXXXXXX I have configured the oh323.conf following: gatekeeper=XX.XX.XXX.XXX alias=XXX-XXX-XX-X Isx the alias equal to the h323id ? And how i have to make a call with the dial app ? I have following config: exten