[Asterisk-Users] iax2 show netstats

10 40 0 0 00 0 -- thanks, yusuf ___ --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] how to communicate two PCs on LAN with

make install. For asterisk do: make make install make samples. (Read the install docs) go to voip-info.org. if the softphones you download a sip, then configure two sip devices in sip.conf, then put correct extensions to dial in extensions.conf. (Read the sample .conf's) thanks, yusuf

Re: [Asterisk-Users] oh323.conf problem

/asterisk/oh323.conf': Found Illegal instruction [EMAIL PROTECTED] ~]# Hi, I have had the exact same problem last week. I have not yet solved it. So instead I am using ooh323, but would prefer to use oh323. Can anyone help? thanks, yusuf -- This message has been scanned for viruses

[Asterisk-Users] FXO/FXS and E1 in same system

= 1-15,17-31 how will the channel numbers change with two cards. thanks, yusuf ___ --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] oh323 - cant load module

the wrong versions of pwlib,h323? -- thanks, yusuf ___ --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] chan_h323 problem

= no noH245Tunneling = yes or noH245Tunneling = no yusuf ___ --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] SIP Realtime Users

Douglas Garstang wrote: Trying to get SIP realtime working here... I'm connected to the database... *CLI realtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI realtime load sipusers

Re: [Asterisk-Users] List of transcoding combinations

Robert Webb wrote: Is there a list or matrix somewhere that shows what codec can be transcoded? I am playing with different allowed codecs between my asterisk box and some of my providers testing voice quality and bandwidth usage on my cable connection, and I occassionally run into an issue

Re: [Asterisk-Users] Creating a voip network... use asterisk?

in a ISDN card, you just put in this route/extension in Asterisk. Anything I missed? -- thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

Re: [Asterisk-Users] How to transmit Video

RAHEEL HASSAN wrote: please tell me that what sip based softphone will beused with Asterisk so that i can trasmit and receive video on my LAN . Yahoo! Mail Hi, i have used eyebeam exten for video. However it is not

[Asterisk-Users] carry forward uniqueid

it is 'unique'. I just want to match a call on different asterisk servers from the cdr's. thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

Re: [Asterisk-Users] ZT_CHANCONFIG failed on channel 2: , Guidance requested

zaptel [zaptel.conf] fxoks=1-2 fxsks=3-4 [zapata.conf] signalling=fxo_ks channel =1-2 signalling=fxs_ks channel =3-4 modprobe zaptel modprobe wctdm hope this helps yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

[Asterisk-Users] Asterisk + SE Linux

Hi guys, I am busy planning to implement SE Linux on my asterisk box. Either that or I will use AppArmor from Suse. I just want to know what are others experiences/incidents with SE Linux or AppArmor thanks, yusuf ___ --Bandwidth and Colocation

RE: [Asterisk-Users] Two PBX

have not over simplified it :) yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

Re: [Asterisk-Users] my zap channel not ringing

span=3,3,0,ccs,hdb3,crc4 span=4,4,0,ccs,hdb3,crc4 bchan = 1-15, 17-31, 32-46, 48-62, 63-77, 79-93, 94-108, 110-124 dchan = 16, 47, 78, 109 yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

RE: [Asterisk-Users] my zap channel not ringing source from

as advised. goksie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: Thursday, March 02, 2006 6:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] my zap channel not ringing ADEGOKE ARUNA

Re: [Asterisk-Users] wake up calls

this event to happen, make a cron entry to move this file into /.../outgoing yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com

[Asterisk-Users] VAD, CNG, for Zap

would greatly appreciate any pointers/suggestions. thanks, yusuf ___ --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] H.323 ( HW PBX to *)

Hi, I'm trying to connect * to Nortel BCM 50, This PBX use H.323 v3 to interface with other PBX. The port use to connect is TCP 1720 but I can't configure this port on my * box. I'm using a H.323.conf file sample to activate the port but the * isn't listening there. Somebody

Re: [Asterisk-Users] Asterisk, SIP phone , NAT

Paul Lacatus wrote: I have an asterisk pbx conected to internet. I need to connect to asterisk a sip phone over the net that will be connected to the internet over a NAT router. I found on the net that SIP is not very NAT friendly. If somebody has a how-to for my problem please share it with

Re: [Asterisk-Users] Asterisk Topology

numbers to other asterisk servers. One question, can each of the local @ servers 'see' each other, or can they only 'see' the main server, because this will change your dialplan. yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk

Re: [Asterisk-Users] Problem with T1 installation

Are you sure the signalling is right, play around with other signalling types and see. also play around with crc on and off Anthony Rodgers wrote: Are you sure you're supposed to be using EM? On Feb 24, 2006, at 5:39 AM, Nitin Joshi wrote: Hi All, I have installed a Digium TE110P card on

Re: [Asterisk-Users] Recommended rack-mountable server anyone?

Hi, we have used about 6 Intel rack mounted servers, with dual Xeon processors, (I have forgotten the exact make.) we used them with Digium quad span pri cards, and some with Sangoma cards, works well . These servers are still pretty solid. they come with two onboard NICs, and with the Digium

Re: [Asterisk-Users] Cannot see the caller id , When calls made from one server to another

=secret host=192.168.20.32 context=project extensions.conf (20.99) contains exten = _3XXX,1,Dial(IAX2/johna/${EXTEN:1},30,r) exten = _3XXX,2,Congestion John, in iax.conf, there are settings for callerid, so try callerid=johna or callerid=asrecieved and usecallerid=yes should do it yusuf

Re: [Asterisk-Users] Problema calling from elesign h.323 to iax

Hi, i'm using an elesign voip gateway esc1700 to call to one iax sofphone (idefisk) and sip device (sipura spa-2002), the trouble es when I make the call using the esc1700 the communication is dropped, this is the log portion: Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Call accepted by

Re: [Asterisk-Users] ~1 sec delay from callee answering to call established on dialout

John Morris wrote: Hi, I've been using Asterisk (now version 1.2.4) for quite a while, and I'm trying to switch from POTS lines to a VOIP termination service. I've had this problem with a few services I've tried, so the problem must be on my end. Here's what I see: When dialing out, I hear

Re: [Asterisk-Users] GSM GATEWAY

Dumpolid Exeplish wrote: Hi everyone, Can anyone give me suggestions on any equipment that can connect from VOIP to a GSM gateway (channelbank that can load up to 30 sim cards and make 30 VOIP-to-GSM calls simultaneously), i hav looked at 2z's Stargate (which has a VOIP card) but the device

Re: [Asterisk-Users] GSM GATEWAY

yusuf wrote: Dumpolid Exeplish wrote: Hi everyone, Can anyone give me suggestions on any equipment that can connect from VOIP to a GSM gateway (channelbank that can load up to 30 sim cards and make 30 VOIP-to-GSM calls simultaneously), i hav looked at 2z's Stargate (which has a VOIP card

Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning

on my side thanks, yusuf ___ --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] realtime, iax, trunk

Ben Dinnerville wrote: Hi All, Is there some way to verify that a channel is using iax trunking? The reason i ask is that i have a scenario where 1 Asterisk system is communicating with another over IAX. System A is using static configuration from the standard files, System B is using

Re: [Asterisk-Users] asterisk h323

directoryu and compoile ooh323. Also has worked well. But this one, as of 1.2.4, only has support for ulaw and alaw and gsm i think. CVS (or SVN) has now support for g72* both are configured very similiar in oh323.conf and ooh323.conf respectivley yusuf

Re: [Asterisk-Users] iax2 trunking known problems?

trixter aka Bret McDanel wrote: On Thu, 2006-02-16 at 14:29 +0200, Zoa wrote: The trunks were made to be maximum 60 simultaneous channels iirc. I doubt seriously you will be able to do 600 simultaneous on any system. (with or without trunking). (at least out of the box). Zoa At 100 with

Re: [Asterisk-Users] asterisk-1.2.4 + asterisk-addons-1.2.1 for mysql realtime

Kamran Ahmad wrote: hi i am using asterisk-1.2.4 + asterisk-addons-1.2.1 on 2.6 kernal. i have added user in sip_buddies and followed http://www.voip-info.org/wiki-Asterisk+RealTime+Sip but my ip phone is not registring properly. asterisk is just sending SIP/2.0 404 Not found. i think it must

Re: [Asterisk-Users] G723 error

or something right before the dial? (To my knowledge there isn't). On 2/15/06, yusuf [EMAIL PROTECTED] wrote: I am assuming you made a profile in sip.conf like so [sipdevice] type=peer host=x.x.x.x ... . . disallow=all allow=ulaw and in extensions.conf exten = _X.,1,Dial(SIP/sipdevice/${EXTEN

Re: [Asterisk-Users] asterisk h323

version of the above also what does config.log say? leonimar cape wrote: Hi Yusuf, I need your info about the installation of the oh323, I have a problem compiling the pwlib. this is the error that I got.. Any ideas, ./configure checking for g++... no checking for c++... no checking for gpp

Re: [Asterisk-Users] G723 error

I am assuming you made a profile in sip.conf like so [sipdevice] type=peer host=x.x.x.x ... . . disallow=all allow=ulaw and in extensions.conf exten = _X.,1,Dial(SIP/sipdevice/${EXTEN}) then this MUST work. :) you can do a sip debug or set debug 10 yusuf Matt wrote: Hi, How do I specify

Re: [Asterisk-Users] CDR for Inbound Calls

handle inbound calls, put it into a certain context, so you can seperate from the rest. yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

Re: [Asterisk-Users] Different Voice Prompts at Different Times

-incoming,s,1) [Ecn-incoming] exten = s,1,Answer() exten = s,2,Background(ecn-welcome) ;exten = s,2,Background(ecn-holiday) exten = s,3,Hangup [Ecn-afterhours] exten = 5000,1,Background(after-hours) yusuf ___ --Bandwidth and Colocation provided

Re: [Asterisk-Users] Developing a call centre app. Communication with asterisk?

. There is java classes that talk to the manager interface, that can pick up any call events, which will allow you to pick up, transfer , answer. yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

Re: [Asterisk-Users] Asterisk errors configuring for PRI

[EMAIL PROTECTED] wrote: Hi, I've just compiled asterisk and get errors when trying to run it. Am I right in thinking that if my digium card is not plugged into our ISDN line then this is normal. If I replace zapata.conf with a blank zapata file asterisk runs fine. The reason I ask is we

[Asterisk-Users] 0h323 - one way audio

. Channel OH323/[EMAIL PROTECTED] (call 'ip$localhost/13673') TX byte count is 5000. Channel OH323/[EMAIL PROTECTED] (call 'ip$localhost/13673') RX byte count is 8000. Channel OH323/[EMAIL PROTECTED] (call 'ip$localhost/13673') do have any ideas? thanks, yusuf

[Asterisk-Users] RE: Wildcard TDM2400P: comments

We are also looking for analog port for fax and dialup modem. Yusuf, would you pls descript what stability issues you had with TDM400P? We are thinking about using TDM400P or Voicetronix OpenPCI-8S. Cheers, Isaac Well, we used the TDM400P a while back, obviously things might have been fixed

[Asterisk-Users] Wildcard TDM2400P: comments

about this new TDM2400P??? thanks, yusuf ___ --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] Re: Ringtone when dialing

yusuf wrote: Hi all, can anyone tell me when (or how) * starts generating ring tone when a call is made. The reason I ask this is I have an E1 coming from a PBX into my * box (CVS 19/07/2005). I have some intermittent problems. 1. Sometimes no ringtone is generated, so I dial a number

Re: [Asterisk-Users] Realtime Replication of a Single File

I think Ryan is right. We have been using Realtime + mysql. We also run reports in realtime that query the mysql db, particularly the cdr table. Al this runs in realtime, which is very important for reporting. It not very cpu, mem intensive. yusuf This sounds like a prime candidate

[Asterisk-Users] Ringtone when dialing

Hi all, can anyone tell me when (or how) * starts generating ring tone when a call is made. The reason I ask this is I have an E1 coming from a PBX into my * box (CVS 19/07/2005). I have some intermittent problems. 1. Sometimes no ringtone is generated, so I dial a number, and the person

[Asterisk-Users] ERROR utils.c:509 tvfix:

These errors just started showing on asterisk cli. Me setup is a pri/e1 card, connected to a philips PBX. google gave no answers. Any ideas??? 2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative timestamp -267121.-532000 2005-11-15 11:25:58 ERROR[23932]: utils.c:509

[Asterisk-Users] Re: How to specify when to go to 102 priority

Steve Davies wrote: On 31 Oct 2005, at 08:25, yusuf wrote: Hi all, I currently have this configuration. exten = _X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5060) exten = _X.,102,Set(PRI_CAUSE=42) exten = _X.,103,Hangup() I have an Asterisk box connected via E1/PRI to Siemens PBX

[Asterisk-Users] How to specify when to go to 102 priority

priority. But it does not go to the 102 priority on ' 488 Not Acceptable Here' . So my question is how to I make it go to priority 102 whenever it fails in priority 1 Any ideas?? thanks yusuf ___ --Bandwidth and Colocation sponsored by Easynews.com

[Asterisk-Users] CHANNEL HANGUP ASSISTANCE

Hi I have encountered a problem with my asterisk. Here is my set-up , I am using E1_PRI as signalling over a Nortel PABX. What i intended on doing is sending a call rejected signal . I have it set-up as PRI_CALLED=21 , it sends the signal but then it hangs up the channel , i need help sending

[Asterisk-Users] Delay in dial

Line 6 has executed, only then does it dial Can anyone help yusuf ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] BAD echo problems with Sangoma and, Telstra

We have an Asterisk 1.0.9 machine with a Sangoma A101 card fitted, it is connected to a Telstra OnRamp E1 in Melbourne, Australia. The problem we are experiencing is extreme echo and clicking noises. These are only audible to the calling party, e.g. the person calling in from the PSTN to

[Asterisk-Users] Re: Delay in Dial

Hi, Try taking out Answer in your extensions.conf. You don't need to answer before dialing a SIP channel. cheers. AK On 9/28/05, yusuf [EMAIL PROTECTED] wrote: Hi all, I am using Asterisk CVS, and I am getting a huge delay in dialing SIP. This Asterisk box is taking calls from a PABX over

[Asterisk-Users] Sirrix bri card:killing the machine

Hi all, I have a sirrix bri card, connected to 2 ISDN lines.I used to get a lot of slips on it, so i put the 'master' setting to 'yes'. But every couple of hours the machine completly hangs, and i have to reboot it. I only get 1 or 2 slips,I dont know whats wrong? yusuf

Re: [Asterisk-Users] Help me how to listen voicemail with SIP 7960

In your settings button 1stly unlock config.Then find SIP Configuration (option 4) after selecting that go to option 7 that is Messages URI. Now edit or put the extension. Thank you Yusuf On 8/11/05, Lokesh kumar [EMAIL PROTECTED] wrote: Hi, Everybody I am running asterisk successfully, I am

Re: [Asterisk-Users] Blank CIDName or CIDNum = asterisk

Another thing you can do. In your dialplan you can define cid in incoming call like this.. exten = s,1,SetCallerID(Unavailable) Thanks yusuf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo

[Asterisk-Users] Problem with realtime SIP

in extensions.conf: switch = Realtime/[EMAIL PROTECTED] to tell asterisk to go to the database to look for the users rofile and extensions yusuf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] asterisk cisco 7960 softkeys [Virus checked]

I need the same for Cisco 7960 with sip image. I need to rearrange the softkeys of the phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Re: telecomFM CellRoute GSM with Asterisk?

I am replying to me!!! I am extremely sorry guys for asking that stupid question!! My problem is solved. I connect that telular to the fxo port of my pbx and nothing else was required. Thank you all. ___ Asterisk-Users mailing list

[Asterisk-Users] Fax detection on isdn

Hi all, I have previously got fax working on a digium card, now im using a sirrix isdn card. The problem is that it cant detect when it is a fax. So how do i get fax detection working on this sirrix isdn card. thanks, yusuf ___ Asterisk-Users

Re: [Asterisk-Users] Remotely Access an Extension

feature is basically forward the call to another number(May be my concept is still not clear!). Person X might be at Y's desk,and wants to login Y's phone with his(X) extension.So I can't use call forwarding. Regards Yusuf On 7/19/05, C F [EMAIL PROTECTED] wrote: On 7/19/05, Yusuf Iqbal [EMAIL

[Asterisk-Users] Remotely Access an Extension

in a distant location from my desk for a long time I could be able to get my phone with me. Thank you, Yusuf Iqbal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE

[Asterisk-Users] telecomFM CellRoute GSM with Asterisk?

I have a telecomFM CellRoute GSM. I want to route call to the cell phones with that device. Has anyone experienced in connecting an Asterisk pbx to telecomFM CellRoute GSM successfully? Please help me to integrate this device with Asterisk Server. Thanx in advance, Yusuf Iqbal

Re: [Asterisk-Users] Cisco 7960s and skinny

Julien, Thanks for your suggestions. You are working on it right? Could you please inform us about your updates? So that we could get those buttons working properly and console become clean! On 6/2/05, Julien Goodwin [EMAIL PROTECTED] wrote: On Thu, Jun 02, 2005 at 01:56:09PM +0600, Yusuf

Re: [Asterisk-Users] Cisco 7960s and skinny

7910's with Chan_sccp are running well except some cases. As usually Transfer, Msgs, Conf,Forward,Speed buttons are not working! Infact I have already knew that, those buttons will not work (from mailing list). It seems like the * console is very busy with messages constantly on it from the sccp

Re: [Asterisk-Users] Cisco 7960s and skinny

Hi Andy, Thank you so much for your help. My 7910's are now working!!!:) Now I can work with those IP phones. I am still monitoring them. I will let you know the further status. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Re: MoH: mgp123 problems

: [EMAIL PROTECTED] Content-Type: text/plain Can you post your conf file for the musiconhold??? Sounds like you haven't defined a default class / context - I could be wrong -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: Wednesday, May 25, 2005 8

[Asterisk-Users] MoH: mpg123 problems

Can you help, Thanks yusuf ___ 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] Cisco 7960s and skinny

Hi Andy, I have been trying to get 7910's work with *. I have tried with both skinny and chan_sccp. Could you please instruct me about the configuration? I have found some detalis about 7920 with sccp in voip-info.org. But I haven't find any document for 7910. Please help me to get them work

Re:[Asterisk-Users] Problem with X101P

Scott Stingel wrote: Some questions: What country are you in? Is there anything else connected to the line from the PSTN? It sounds like you have a marginal condition, such as insufficient loop current perhaps. Do have any features, such as call waiting, on the line? Do you know how far

[Asterisk-Users] Problem with X101P(Red Alarm)

I have bought some Wildcard X101P and Generic Clones for my Asterisk PBX. Now I can place and get calls through the lines/channels. Everything is okay but the problem is when I call outside through our PSTN line, after few minutes the connection breaks down. The same thing happens in case of

[Asterisk-Users] problem with skinny

I have a couple of Cisco 7910's and I'd like to get them working with asterisk. I have two X101P wild cards installed and they are functioning well (Other two cards showing Red Alarm after few minutes conversation). I have configured foure cisco 7960 with SIP and they are working fine with *.I

[Asterisk-Users] help:Memory Consumption

Can anybody help me to figure out how much memory per minute is consumed for voicemail applications? And how many concurrent calls can be handled at a time in Asterisk? so that, I can choose the specification for Server to setup Asterisk for a large number of users.

[Asterisk-Users] Re: Asterisk-Users Sip Reload or Realtime

a realtime load familyname columnname searchstring in CLI Doug yusuf ___ 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

[Asterisk-Users] Problem with X101P

Previously I have posted the same mail but no one answered me...Sorryfor resending the mail.I have bought a Wildcard X101P for my Asterisk PBX. Now I can placeand get calls through the lines/channel. Everything is okay but theproblem is when I call outside through our PSTN line, after fewminutes

[Asterisk-Users] Help: Problem with X101P

I have boughta Wildcard X101P for my Asterisk PBX. Now I can place and get calls through the lines/channel. Everything is okay but the problem is when I call outside through our PSTN line, after few minutes the connection breaks down. The same thing happens in case of incoming calls. I have

RE: [Asterisk-Users] zoom x5v and *

and configure the modem couple of times to make it register to asterisk. Yusuf Alakavuk Grid Bilisim Teknolojileri A.S. Kustepe Mahallesi Leylak Sokak Murat Is Merkezi A Blok Kat:2 Daire:9 34387 Sisli Istanbul Türkiye Tel : +90 (212) 336 92 55 Fax : +90 (212) 266 25 50 -Original

RE: [Asterisk-Users] Voicemail problem

turning off VAD and silence suppression at the client can solve this problem. Yusuf Alakavuk Teknik Danman - Technical Consultant Grid Biliim Teknolojileri A.. Kutepe Mahallesi Leylak Sokak Murat Merkezi A Blok Kat:2 Daire:9 34387 ili stanbul Trkiye Tel : +90 (212) 336 92 55 Fax

RE: [Asterisk-Users] RE: Voicemail problem

Hi, We experinced this problem with cisco IP phones and it was solved after turning on silence suppression . I don't know any other way than this. Sorry. Yusuf Alakavuk Teknik Danisman - Technical Consultant Grid Bilisim Teknolojileri A.S. Kustepe Mahallesi Leylak Sokak Murat Is Merkezi

RE: [Asterisk-Users] X-lIte behind NAT and Asterisk behind NAT

all you have to configure your X-lite's network parameters to use your clients external ip address for your SIP communication. After all it will be working if you have further problems you can read the documents at the http://www.voip-info.org site by searching SIP NAT Yusuf Alakavuk Teknik

RE: [Asterisk-Users] Cisco Wireless IP Phone 7920

You can use it with SCCP channel. We have tried and it works well. But I don't recommend this phone it's battery is not enough. It has a 3-4 hours time of standby. Best Regards. Yusuf Alakavuk Teknik Danisman - Technical Consultant Grid Bilisim Teknolojileri A.S. Kustepe Mahallesi Leylak

RE: [Asterisk-Users] Call pickup

Hi, Have you configured features.conf file? the line which enabled call pickup is commented and you have to un comment the line for call pickup to work. Also you can define the numbering for call pickup there Thanks. Yusuf Alakavuk Teknik Danman - Technical Consultant Grid Biliim

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