similar to: Dial Commands "D" Option Question

Displaying 20 results from an estimated 4000 matches similar to: "Dial Commands "D" Option Question"

2005 Jul 27
2
oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6
Abwesenheitsnotiz: [Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6Hi All I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb ram, with g729 for i686 , (fedora 1). my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen otherparty realtime voice , but other party geting sip party's voice 1 sec later (not
2005 Jul 19
3
CID Matches On Incoming BroadVoice???
I have been trying to make Broadvoice match incoming Caller ID and do specific things based on the number received, but due to Broadvoice requiring the "s" to start off an incoming extension, I cannot get this to work. Has anyone been able to do this? Here are some examples of my setup: from sip.conf: context=broadvoice-incoming from extensions.conf [broadvoice-incoming] exten =>
2005 Aug 16
3
ASTCC astcc-config.conf card length question
I currently have my astcc databases card lenghts at 7 digits long. I would like to expand this to 10 digits now though. Will I screw things up if I leave the old 7 digit long pins in there and start using/generating 10 digit pins? Thanks
2005 Oct 12
2
Broadvoice Outages?
I've been having a lot of problems with Broadvoice lately. Anyone else been without service for extended periods of time this week?
2004 Aug 11
7
H323 call dropped when answered
Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.
2006 Oct 15
3
VoicePulse Connect 4 Channel Limit?
Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. I noticed that they supposedly only allow 4 channels for free and then you have to pay $20 a month extra per channel. I'm guessing this is for inbound
2005 Jul 27
2
Regarding Call Hold
> Hi All, > > We are using asterisk for testing our home gateway setup. > We have implemented Call Hold feature in our application. > In our Application we have written code in such a way that for an INVITE > for > putting a SIP phone on HOLD will contain connection address "0.0.0.0" in > the SDP message. > We expect the same connection address i.e
2008 May 15
3
[PATCH 1/4] ocfs2: Fixes pipe_buf_operations->pin switch to confirm in 2.6.23.
Signed-off-by: Tiger Yang <tiger.yang at oracle.com> --- Config.make.in | 1 + configure.in | 6 ++++++ fs/ocfs2/Makefile | 4 ++++ kapi-compat/include/pipe_buf_operations.h | 10 ++++++++++ 4 files changed, 21 insertions(+), 0 deletions(-) create mode 100644 kapi-compat/include/pipe_buf_operations.h
2006 Jul 17
10
String manipulation and formatting
I'm trying to write a simple function that does the following: [command] xify(5.2) [output] XXX.XX [command] xify(3) [output] XXX Any simple solutions (without using python/perl/unix script/...)? Thanks, Saghir --------------------------------------------------------- Legal Notice: This electronic mail and its attachments are i...{{dropped}}
2006 Oct 25
3
Quintum DX as gateway to PSTN for Asterisk
Hello, I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial number which is connect to Quintum, and call is diverted to *. I don't know what I should set, if I want call from SIP_phone registred in Asterisk to PSTN via Quitnum. I set in sip.conf account for Quintum [sip_proxy-out] type=peer outboundproxy=QUINTUM_IP , and changed extensions.conf. When
2006 Jun 25
5
FW: Asterisk Quintum A800 SIP Mode
Hello, I got Quintum A800 with the P5-2-1 firmware. I configure my asterisk trunk as followed: [SIP_BD1] type=peer qualify=yes host=192.168.0.254 disallow=all context=from-pstn allow=h723 and inside the quantum I change the config sip useragent to 5060. Up to this part if I run sip show peers, I got: asterisk1*CLI> sip show peers Name/username????????????? Host??????????? Dyn Nat ACL
2009 Jan 14
15
Backport patches to ocfs2 1.4 tree from mainline
Found 15 patches (out of 162) that appeared relevant to ocfs2 1.4. Please review. Sunil
2006 Mar 21
2
need to make my oh323 work with quintum no gatekeeper
Hi all, Can someone share with me his experience in making asterisk-oh323 talk to quintum gateway without gatekeeper. My set up is QUINTUM GATEWAY ------IP----M ASTERISK (OH323) Both are gateways.. but I don't know what authentication I will set up in oh323.conf and how to set it up I will be glad if anyone can help Goksie
2005 Mar 18
15
Meetme2 compilation problem
Hi All, I am trying to compile meetme2 in my asterisk box and getting the following compilaton error. Please help me to sort it out. cc -fPIC -c -o app_dial.o app_dial.c In file included from app_dial.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function)
2003 Jun 07
4
SIP, NAT & Asterisk
Hi all, -------- beacause I am a newbie in the asterisk ralm and the existing documentation could not satisfy I'd like to ask you some Questions: 1. Does somewhere in the Internet exist additional documentations for asterisk configuration ? 2. Does Asterisk work as a standard SIP Proxy ? 3. I am just installing a Asterisk PBX in our institute and additionally I purchased some ot the Snom
2006 May 23
1
Quintum Tenor DX 3020 problem to register on Asterisk
Hi, I'm having problems to register Quintum Tenor DX 3020 on a Asterisk box with SIP. Asterisk always returns "Username/Password mismatch". I've tried all configurations that was on the Quintum's manual, but no success. I've tested the same username and password with a Linksys (PAP2-NA) with the same asterisk box, and it worked fine. Where is the problem ?
2009 Jan 10
3
Asterisk/GXW410x IP Analog Gateway
Hello All, I am trying to setup a small system where Nextone Softswitch will send traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog Gateway but for some odd reasons the call are flashed back from Grandstream to Asterisk and creating a Black loop... I did follow the instructions provided by Grandstream support but it doesn't seems to be working...
2004 Aug 23
1
Asterisk <------- Quintum SIP Registration
Hi All I'm trying with no luck to connected the Quintum D series Gateway with the new SIP release to asterisk. Have anyone done this? If yes then how should I configure the sip.conf to accept the registration? maybe a sample config? Thanks /Krystian
2006 Apr 05
6
transforming g729 files to wav files
Hello list, is there any open-source software that recodes g729 sound files to wav sound files ? The only way (at least) to do such transformation is with interactive form on: http://www.asteriskguru.com/audio_conversion.php Tofik Suleymanov
2007 Mar 08
4
Asterisk distributed deployment
Hello all, I post this issue thinking too that could help other people on an asterisk deployment over distributed offices considering both quality, prices, devices and so. Well, i am working on a deployment of a telephony system based in asterisk. My company have a central office with seven remote offices connected all through a VPN. To reduce and evaluate costs i consider solutions like: