Re: [asterisk-users] Experiences with grandstream GXW 4024 FXS?
I have 4 instalations with with it, the first one from agust, and witout incident, in this time we make only 1 restart. Regards Mehdi Chouikh http://www.voz-ip.info http://www.unitelexperts.com http://www.mitelefonovirtual.com On Wed, Dec 24, 2008 at 6:05 PM, Daniel Hazelbaker dan...@highdesertchurch.com wrote: I use the GXW-4008 and have never had any problems with it. Right now it runs 3 analog phones, but we were using it to link our old NEC phone system to the new Asterisk system, so it was used quite a bit and never once had an issue. Daniel On Dec 24, 2008, at 5:30 AM, Hector Quiroz wrote: HI all, does anyone already implemented the GXW-4024 FXS? Some distributors doesn't recommend it for high volume operations. regards, Hector. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mehdi Chouikh http://www.voz-ip.info http://www.unitelexperts.com http://www.mitelefonovirtual.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk access Postgres for Realtime Configuration
Yes you can use res_conf_pgsql.so is present in asterisk 1.4 On Oct 7, 2006 1:22 AM, John Miloo [EMAIL PROTECTED] wrote: Hello Comunity, How can I get Asterisk realtime working with Postgres? (without ODBC)? Thanks John /doc/realtime.txt in Version 1.4 Beta2 Currently there are three realtime database drivers: * ODBC: Support for UnixODBC, integrated into Asterisk The UnixODBC subsystem supports many different databases, please check www.unixodbc.org for more information. * MySQL: Found in the asterisk-addons subversion repository on svn.digium.com * PostgreSQL: Native support for Postgres, integrated into Asterisk ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mehdi http://www.voz-ip.info ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 SIP Jitter Buffer
I have asterisk 1.4.14 in 3 of my 8 servers for 3 weeks on productions systems, but i had problem with adapative jitter buffer, when i active it there are no sound. Regards On Nov 6, 2007 9:16 PM, Gregory Boehnlein [EMAIL PROTECTED] wrote: Are you running the SIP Jitter Buffer? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Luc Moreira Sent: Monday, November 05, 2007 10:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 SIP Jitter Buffer Gregory We have many Asterisk 1.4.13 in production solid like a rock. Couples examples: a) Asterisk 1.4.13 + Unicall + 2 E1 MFCR2 Digium + Legacy PBX 60+ Extentions / IVR / 10~30 concorrent calls b) Asterisk 1.4.11 + 1 E1 ISDN PRI Digium 50+ Extentions / IVR / 5 Queues / ~2000 call/day c) Asterisk 1.4.13 + 4 E1 ISDN Digium (working in progress) CallCenter / 150 PAs / 15 Queues / expected 8000 calls/day -- Luc Gregory Boehnlein escreveu: Hello, I'm running into a few situations on lossy network links where a SIP jitter buffer w/ some PLC would be helpful. My main TDM gateways are running 1.2 (which is solid, stable, reliable and very very very well behaved when you know it's limitations), but I'm considering upgrading them before the end of the year to 1.4. Two of the main reasons that I would do this are Variable Length DTMF and SIP Jitter Buffering. I would be very interested in hearing from anyone that is actually running 1.4 in a PRODUCTION environment, gatewaying SIP to TDM using Digium cards. To me, production means being able to have 3-4 PRI circuits maxed out for 12+ hours a day and 7+ call setups / second. Anything less than that is not really going to be an accurate comparison to what I have running. Anyone have any feedback about this combination? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by N2Net Mailshield, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with conferences, Vlada, Pancevo
the forst problem you have, you need to los the meetme module, and second one is a timer, for that you can use ztdummy, compiling the zaptel driver. Regards On 5/7/07, Ronaldo [EMAIL PROTECTED] wrote: Hi, I'm not sure, but MeetMe needs some timer module from zaptel project. Try read about timers for MeetMe application. Ronaldo. Vladimir Kovacevic wrote: Hi, I have problem with setting up a conferences. When I dial the reserved conference number from xlite the line gets hunged up and on a console there is a following message: WARNING[5924]: pbx.c:1783 pbx_extension_helper: No application 'MeetMe' for extension (internal, 1234, 3) exten = 1234,1,Answer() exten = 1234,4,MeetMe(1234|Md) exten = 1234,101,HangUp() meetme.conf: [general] [rooms] conf = 1234 What I did wrong? Thx, Vlada ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Suggestion for 2 PRI with call recording
We have a similar system up and runing for 6 months, wiith 60 channels, and average of simultaneas recorded calls us between 20 and 30. We make test for recording 60 calls without any problems We use a PIV Dual core with 3.2 Ghz with 2 mb of cache and 1Gb Ram. regards Mehdi On 12/13/06, A.R. Nasir Qureshi [EMAIL PROTECTED] wrote: Hi everybody, I intend to setup an Asterisk Box to handle 2 ISDN PRI (60) channels, for incoming calls. Callers will first get an IVR, and 20 of them can talk to Agents over IP Phones. The Agent Calls will be recorded. Thus at max, 20 calls will be recorded at a time. Please suggest the hardware I should use. -- Regards, Nasir. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX using T38
Hello Asterisk implement only passtrough T.38, so you cant terminate calls with asterisk using T.38. You need T.38 gateways. Regards On 11/13/06, Ricardo Carvalho [EMAIL PROTECTED] wrote: Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the available patch found at URL http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3 that is announced to support it too. With both Asterisk versions, I've sent with success FAXes between two FAX machines each one attached to an ATA interface, both registered in Asterisk. Although I can't send FAXes to PSTN using T38. The problem, as far as I know, might be assigned with the Content-Length shown in the message header of every SIP message negotiating T38 parameters. I've observed that after leaving Asterisk, the Content-Length of every message carrying T38 parameters gets shorter than truly is, and contrarily to my ATAs that seem to don't care about this, my Telco analyses the packet length written in this messages and truncates them, aborting the call. Does anyone experienced this too? Any ideas besides editing the chan_sip.c code to fix this problem? Thanks, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Moscow Dids
Hello I need Moscow dids urgently, Contact me offline [EMAIL PROTECTED] Regards Mehdi Chouikh Universal Telecom Spain ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] looking for Russia and Israel Dids
hello I am looking for Israel and Russia DiDs. Please email me to [EMAIL PROTECTED] REgards ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 or h323
Hello Personaly i prefer oh323, i am using for one year whitout problems. and is more easier to configure. regards On 9/1/05, Steve Ducat [EMAIL PROTECTED] wrote: I have just signed up for 2 landline numbers in China. They haveoffered to sell me 2 h323 compatible handsets which I have declined as I want these numbers to ring into my * box.They have given me the following info (modified for security)..Protocol = H323Gatekeeper = 210.21.118.xxxH323ID = .HMA0200.10szxn-hxxxe164 = 02022xx2912 H323ID = .HMA0200.10szxn-kxxxe164 = 02022xx2913Really what I want is for * to act as the endpoint.So the big question, do I use oh323 or h323 or something else. I amall confused about who is the gatekeeper, who is the gateway. I just want * to register with the gatekeeper so they will pass * all theincoming calls.Which one do I use and how would I tackle the conf file to registerwith the gatekeeper.Any help would be appreciated. Steve Ducat.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_dial.c:977 dial_exec_full: Unable to createchannel of type 'Zap' (cause 0)
The clone work good, Try to change your dialplan for zap channel and it will
work.
Try this exten =_11.,1,Dial(Zap/1/${EXTEN:2},90,Tt).
It work for mi with my clone card and my x100p (original card) for long
time.
saludos
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, June 15, 2005 7:48 PM
Subject: Re: [Asterisk-Users] app_dial.c:977 dial_exec_full: Unable to
createchannel of type 'Zap' (cause 0)
I don't have clone card to verify this, but I think you'll find the
chipset on that particular card is not the same chipset used on the
digium card. Since the asterisk drivers are written for specific
chipsets, I'd have to suggest you've got an almost zero chance of
making the clone work.
No Ideas? This seems like quite a common issue but I have searched and
searched for a solution and not found any?
Cheers.
Sandy.
Sandy Thomson wrote:
Hi,
Ive been struggling with asterisk for a few days now. I understand
pretty much how it works and how to tie things together (SIP - SIP
internally works fine etc), but just cannot get asterisk to work with an
X100P clone (its a Ambient MD3200, if that means anything to you guys).
I have tried (initially) asterisk 1.07 with zaptel 1.07, and now
Asterisk CVS-HEAD with zaptel cvs. Both give me the same error.
/etc/zaptel.conf
--
fxsks=1
loadzone=uk
defaultzone=uk
/etc/asterisk/zapata.conf
--
[channels]
language=en
context=incoming
signalling=fxs_ks
channel = 1
/etc/asterisk/extensions.conf
--
[general]
static=yes
writeprotect=no
[local]
exten = _11.,1,Dial(Zap/1,9w${EXTEN}:2)
[incoming]
exten = s,1,Answer()
exten = s,2,BackGround(demo-congrats) ; Play a congratulatory message
[outgoing]
exten = _9.,1,Ringing
exten = _9.,2,Wait,2
exten = _9.,3,Answer()
exten = _9.,4,Dial(ZAP/1/9w${EXTEN}:1)
[default]
include = outgoing
Loading zaptel modules:
--
asterisk zaptel # modprobe zaptel
asterisk zaptel # modprobe wcfxo
asterisk zaptel # lsmod
Module Size Used by
wcfxo 12576 0
zaptel222916 1 wcfxo
crc_ccitt 1952 1 zaptel
asterisk zaptel # ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
Running asterisk
--
asterisk zaptel # asterisk -gc
Registering builtin applications:
[AbsoluteTimeout]
== Registered application 'AbsoluteTimeout'
[Answer]
== Registered application 'Answer'
[BackGround]
== Registered application 'BackGround'
[Busy]
== Registered application 'Busy'
[Congestion]
== Registered application 'Congestion'
[DigitTimeout]
== Registered application 'DigitTimeout'
[Goto]
== Registered application 'Goto'
[GotoIf]
== Registered application 'GotoIf'
[GotoIfTime]
== Registered application 'GotoIfTime'
[ExecIfTime]
== Registered application 'ExecIfTime'
[Hangup]
== Registered application 'Hangup'
[NoOp]
== Registered application 'NoOp'
[Prefix]
== Registered application 'Prefix'
[Progress]
== Registered application 'Progress'
[ResetCDR]
== Registered application 'ResetCDR'
[ResponseTimeout]
== Registered application 'ResponseTimeout'
[Ringing]
== Registered application 'Ringing'
[SayNumber]
== Registered application 'SayNumber'
[SayDigits]
== Registered application 'SayDigits'
[SayAlpha]
== Registered application 'SayAlpha'
[SayPhonetic]
== Registered application 'SayPhonetic'
[SetAccount]
== Registered application 'SetAccount'
[SetAMAFlags]
== Registered application 'SetAMAFlags'
[SetGlobalVar]
== Registered application 'SetGlobalVar'
[SetLanguage]
== Registered application 'SetLanguage'
[Set]
== Registered application 'Set'
[SetVar]
== Registered application 'SetVar'
[ImportVar]
== Registered application 'ImportVar'
[StripMSD]
== Registered application 'StripMSD'
[Suffix]
== Registered application 'Suffix'
[Wait]
== Registered application 'Wait'
[WaitExten]
== Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
== Parsing '/etc/asterisk/modules.conf': Found
[res_features.so] = (Call Parking Resource)
== Parsing '/etc/asterisk/features.conf': Found
-- Registered extension context 'parkedcalls'
-- Added extension '700' priority 1 to parkedcalls
== Registered application 'ParkedCall'
== Registered application 'Park'
== Manager registered action ParkedCalls
[res_musiconhold.so] = (Music On Hold Resource)
== Registered application 'MusicOnHold'
== Registered application 'WaitMusicOnHold'
== Registered application 'SetMusicOnHold'
== Registered application 'StartMusicOnHold'
== Registered application 'StopMusicOnHold'
== Parsing '/etc/asterisk/musiconhold.conf': Found
[chan_zap.so] = (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, FXS Kewlstart
Re: [Asterisk-Users] Hardware Capacity/Configuration
If use Alaw or ulaw as codec, i think it's enough. But if you need to make transcoding to a hard codec like g729, g723, you have to look other cpu. regards - Original Message - From: Daniel Salama [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 04, 2005 4:21 AM Subject: [Asterisk-Users] Hardware Capacity/Configuration I know this is a frequent topic on the list. Sorry if this creates more bandwidth but I couldn't get my specific answer from neither the wiki nor searching the list. I've read that a P4 3GHz+ should be sufficient to handle a 4 T1s on a single CPU machine. I am setting up a proof of concept machine but I was only able to get a P4 1.6GHz machine. If this machine is only going to be forwarding the calls to another, much more powerful, Asterisk machine which will handle more demanding call processing rules, scripts, Monitoring, etc, do you think this CPU will be able to handle the 4 T1s? Will it handle 3? 2? 1? Efficiently, of course. The idea is to setup a basic VoIP gateway whose only intelligence will be to forward ALL incoming calls to another Asterisk box using IAX as well as placing outbound calls through the T1s from other Asterisk boxes communicating using IAX. Comments? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk to analog pbx
Hello all is right, the analog extension should ring, but maybe your dialplan is not correct or you call a bad extension in you PBX. can you post your dialplan?, to see it. regards - Original Message - From: Julio Saura [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 2:37 PM Subject: [Asterisk-Users] asterisk to analog pbx Hi there i have an asterisk box running ok, and now i am trying to integrate it with my local analog pbx So far, i have connected the fxo port of my * to an analog extension port of my analog pbx. As far as i know, if a call an extension of my analog pbx on a sip phone ( i have done the right dial plan for routing these calls to de zap channel ) the analog pbx extension should ring ... am i right? asterisk says the call is done, but the analog extension keeps in silence .. :? any clue, am i doing something wrong? Best regards. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A-Z Termination
Hello I am looking for A-Z termination please send me your prices off-line. Protocols: SIP, IAX Codecs: G723, G729, GSM Regards Mehdi Chouikh Universal Telecom www.unitelexperts.com Tel: +34 902023154 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
