[Asterisk-Users] How many line appearance can Snom 200 handle?

Snom 200 has be set up with extended key pad. The product literature also mention multiple sip registration. How many registration can it handle? It does not seem to appear in the user manual. David Kwok ___ Asterisk-Users mailing list Asterisk-Users@

[Asterisk-Users] How to announce the DNID to the called party

How to announce the DNID to the called party who picks up the phone and say the correct greeting? I suppose it has to say to the called party before the call is bridged. So it has to do something before the dial command transfer the call. Any ideas? David Kwok __

[Asterisk-Users] Phones for vitural office business

I am looking for IP phones that are suited to serviced office operation. The business is to answer calls for customers. The incoming lines are E1 and customers are allocated with DID. So the customers' phone can be answered with the customers' designed messages and instruction. This can be done eas

[Asterisk-Users] OZ ISDN

Kimble Young wrote: "If you go the analogue route: * You'll get poor audio compared to ISDN which is crystal. * Each number will act like a seperate line unlike with an ISDN card where you can receive two calls simultaneously on the same line. * You'll lose cool ISDN features like call deflection.

[Asterisk-Users] Oz ISDN

In Australia, Telstra, the local telco provides isdn modem for isdn connection. The modem has 2 analogue telephone jacks and a serial port for connection to dialup internet. My question is that will it be possible to use Zaptel TDM02B to connect to the analogue jack instead of getting a fritz c

[Asterisk-Users] DID/T1

I need clarification as to DID in T1 connection. T1 provides 24 channels for voice/data. Do it assign each channel to particular DID. Or you can have unlimited DID to share the 24 channel as an example. ie. Outgoing/incoming traffic is not bound to particular channel. Whatever is available will

[Asterisk-Users] illegal instruction - on Via board

I have sorted out the problem of compiling the CVS 040608. When launched the server die with " illegal instruction" error. Although the Makefile in asterisk is changed to PROC=i586. The Makefile in codecs/ilbc still has a line of reference by using uname -m which will come up with i686 architec

[Asterisk-Users] illegal instruction -via c5

I rolled back asterisk cvs to 5/25 and it still runs and aborted with "illegal instruction" I am not too familiar with gdb and not too sure how to trace the illegal instruction. Has anyone have a working cvs version for via hardware? David Kwok ___ Asteri

[Asterisk-Users] change cisco ata 186 dial behaviour

I have ata-186 and grandstream connected to asterisk using sip. I have a voip account with ATP, in Australia. In order to ring HK, I need to dial 0011852. Grandstream behaves normally and send the whole series of digits and it connects ok. But ATA-186 somehow only allow only 11 digits. ON

[Asterisk-Users] illegal instruction

I have just compiled the latest cvs 040605 and have this illegal instruction error when launched asterisk. It is compiled on Via c5 processor. In the asterisk/Makefile I have set PROC=i586 but it does not help the situation.   Any suggestion.   Regards David Kwok smime.p7s Description: S/MI

[Asterisk-Users] Grandstream phone from speaker phone back to handset

I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to

[Asterisk-Users] iax2 trunk - unable to accept trunk packet

My * box which has the zaptel card is also connect to voip termination using zaptel card. The same problem exist as with the * box which does not have the zaptel card. The tech director of the voip provider told me that it is not a timing issue and more to do with my config. Let's not cynical

[Asterisk-Users] iax2 trunk - unable to accept trunk packet

I have 1 * box having x100p installed and the other has no zaptel card at all. both of the * box has compiled in ztdummy module and both have been activated by modprobe ztdummy. When using trunk to connect the 2 *box. The one without zaptel card complaint about unable to accept trunked packet:

[Asterisk-Users] cronjob to reboot gs101

I have used curl to reboot the GS101 as follow: curl -c cookies.txt -d"P2=x&Login=Login&gnkey=0b82" http://192.168.1.xxx/dologin.htm curl -b cookies.txt http://192.168.1.xxx/rs.htm Put these 2 lines in a script and use cron to reboot everyday. -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel

[Asterisk-Users] cron job to reboot GS101

Does any one regularly reboot GS101? It sometimes lost registration with * and needs to be reboot. What is the best way to do it by cron? David Kwok ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-use

[Asterisk-Users] asterisk gui client

I have looked at matt's asterisk gui client at sourceforge. I am not a programmer by trade. The documentation there seems to be a bit lacking. Has anyone have the experience in installing the gui client and may perhaps have a how-to document available for sharing. -- David Kwok Tel: 612 9929208

[Asterisk-Users] RE: message lights and stutter tones

[EMAIL PROTECTED] wrote: --__--__-- Message: 2 Date: Mon, 8 Mar 2004 20:14:00 - (GMT) From: "Simon Chappell" <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: [Asterisk-Users] message lights and stutter tones Reply-To: [EMAIL PROTECTED] Hi al I have 3 GS 101's plugged into asterisk. They w

[Asterisk-Users] iax2 trunk - no matching peers

I have cvsed to the 3/9/04 version. I have not been able to do trunking with other iax2 server. I was able to do it before. What are the procedures to diagnose this problem? Would it be firewall related? Would it depend on both peers with the same cvs version? -- David Kwok Tel: 612 99292086 ex

[Asterisk-Users] flash button on GS101

Has anyone using the flash button on GS101 to access call waiting? My experience is that it does not work. I read in the list that it may need to tweak the flash duration to under 100msec. Has anyone have any solution? -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002

[Asterisk-Users] Hanging GS101 in a upright position

Has anyone tried to hang GS101 phones on a wall? It has recess holes at the back of the base where you can hang it on a wall. What it lacks is that the handset is not supported for this upright position. Has anyone done any modification on it? I was thinking about velco the handset. -- David

[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call

*CLI> Mar 3 12:55:05 WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) This has been brought up in the previous post but it does not seem to have an answer for it so far. I cvs the stable v1.0 this morning after compili

[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2959 - 10 msgs

[EMAIL PROTECTED] wrote: Message: 10 Date: Mon, 01 Mar 2004 10:02:54 -0500 From: John Fraizer <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CVS login Reply-To: [EMAIL PROTECTED] Glenn Dalgliesh wrote: I seem to be having trouble with cvs login. anyone having similar pro

Re: [Asterisk-Users] exit

Just use control-c, you will be able to exist and leaving asterisk continue to run in the background. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature

[Asterisk-Users] Failed to start asterisk

I am using mini-itx motherboard and I installed asterisk stable from cvs. However below is the messages when starting asterisk by safe_asterisk. Anyone spotted the cause of not starting. Last login: Fri Feb 27 10:40:44 2004 [EMAIL PROTECTED] root]# safe_asterisk [EMAIL PROTECTED] root]# /usr/sbi

[Asterisk-Users] RE: Message waiting light not coming on

I cannot get MWI working either with GS101 firmwire 1.0.4.39 My sip.conf has the mailbox number specified. voicemail.conf has mailbox set up. I have collecting mail fine. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature

[Asterisk-Users] system speed dial

have someone implemented some system memory for speed dial? due to the lack of speed dial/memory of GS, I feel the need to introduce some system speed dial function to supplement Gs' deficiency. I am thinking along: recording speed dial exten => _*51.,1,Playback(speed-dial) exten => _*51.,2,Pla

[Asterisk-Users] more codec negotiation problems

I am playing with MeetMe conference. I have 4 Gs101 phones and it does g729A, 711 etc. I have purchase 2 g729 licence. All gs phones are set to use g729. In this case, the system only allow 2 g729 channel for conference. Then I change the GS phone setting to use 711ulaw and still have the same

[Asterisk-Users] codec translation

The route of my call is: gs101-->asterisk-->iaxtel-->asterisk-->gs101 I have 2 g729 from Digium and calls to iaxtel can only be in gsm format. The GS101 phones are set to use g729, then 711ulaw. However when the called GS phone is picked up the connection is terminated. These are the console m

[Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone?

|I cannot call out with my SIP phone though. It'll dial, ring my cell |phone twice and then give up and complain that its busy. Even if I try |to answer the cell phone during the first ring. | |Does anyone have a config they could share with me on how to make this |setup work? This sounds like it s

[Asterisk-Users] Hyuandi pstn handsets

I have 5 Hyuandi pstn handsets and wondering it is possible to plug into tdm400p or ata-286? Perhaps a general question whether handsets from PSTN pbx can be reused with Asterisk? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 ___ Asterisk-Users mai

[Asterisk-Users] RE: "multicasting" conference calls

>>Use the outgoing call feature of asterisk to have the servers join >>each >> others conferences. It's very simple. Sorry, I am not quite sure what is the outgoing call feature. Would you please elaborate a bit. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002

[Asterisk-Users] RE: codec negotiation prob solved

fer=250 maxexccessbuffer=50 register => dkwok:[EMAIL PROTECTED] tos=lowdelay [iax_home] type=friend context=int-ext auth=md5 user=iax_home secret=cc trunking=yes disallow=all allow=gsm host=dynamic qualify=yes [iaxtel] type=friend disallow=all disallow=g729 allow=gsm trunking=yes context=fro

[Asterisk-Users] Echo cancellation

I had problem with echo on my GS phone and it is now resolved. What happens was that the order of settings were incorrect in the zapata.conf. All the echo cancellation settings and other settings to effect on a particular channel have to be set out before the occurence of the channel no. see z

[Asterisk-Users] Re: X100P / Echo / ZTMONITOR CAN2,3, etc.

[EMAIL PROTECTED] wrote: The strange thing here is that asterisk is not removing the echo. I did notice that the 0.7.1 tar did not do the echo cancel very well and Mark suggested that I go back to the CVS, which did wonders. You might also verify that echo cancellation is actually turned on. En

[Asterisk-Users] codec negotiation

I have outgoing connection to iaxtel and another iax server A. iax server A only accept g729 codec while iaxtel is something I am not quite sure of. At the moment iaxtel only accepts gsm. I remember previously it does accept g729. my problem due to the switching between codec when making outgoi

[Asterisk-Users] x100p dropping incoming calls

I have been experiencing hung up when answering incoming calls through x100p. NOTICE[1242768320]: chan_zap.c:4584 ss_thread: Got event 2 (ring/Answered).. -- Executing Wait("Zap/1-1","1") in new stack -- Executing Answer("Zap/1-1","") in new stack -- Executing DigitTimeout("Zap/1-1"."5") in new s

[Asterisk-Users] How can you savage a failed call transfer

I have a couple of cummsy user who always lose a call when the transfer is not done properly ie due to dialing a wrong digit, etc. My question is that is it possible to savage a failed call transfer? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Si

[Asterisk-Users] FWD/Iaxtel/Asterisk codec use

The codec issues with different services and sip phone are the most complicated and trusting experience when using Voip services. I had been able to connect to FWD behind a firewall by using Iaxtel using g729. Just recently, about a week, every time I tried to call FWD, the connection simply ti

[Asterisk-Users] Calling from Iaxtel to FWD users always busy

Just recently my calls through iaxtel to FWD user do not go throuhg due to busy circuit. Wonder if there is any change to the setup of Iaxtel? -- Executing Dial("SIP/1002-246A", "iax2/xxs:[EMAIL PROTECTED]/[EMAIL PROTECTED]") in new stack -- Called xxx:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call

[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2749 - 7 msgs

You can have a look at wiki on iax trunking plus notes on setting up x100p card. David Kwok Message: 5 From: "Maninder Bhatia" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Date: Thu, 5 Feb 2004 16:12:07 -0500 Subject: [Asterisk-Users] X100P - Asterisk - Asterisk - X100P setup help Reply-To: [EMAIL

[Asterisk-Users] Transfer of call from a call queue

I have a call queue set up. The phones are xlite soft phone and Gs101 hard phone. I have AgentCallback setup. After the call is answered by an agent, it cannot transferred to another extension. As soon as transfer is executed, the call is hung up. Is this normal behaviour or I miss something.

[Asterisk-Users] RE: setting up ---- newbie

Date: Sun, 1 Feb 2004 18:24:51 -0400 Subject: [Asterisk-Users] setting up newbie Reply-To: [EMAIL PROTECTED] This is a multi-part message in MIME format. --=_NextPart_000_00A4_01C3E8F0.AED1DC90 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printa

[Asterisk-Users] Call Queues

I have setup AgentCallbackLogin and the agents have been logged in successfully. However when calls are queued and an agent picks up the call. It just hang up the call. On the command console it does say the agent "agent 1001 hang up on customers. they must be pissed off". I agreed. My queues.conf

[Asterisk-Users] Call Queues

I have setup AgentCallbackLogin and the agents have been logged in successfully. However when calls are queued and an agent picks up the call. It just hang up the call. On the command console it does say the agent "agent 1001 hang up on customers. they must be pissed off". I agreed. My queues

Subject: Re: [Asterisk-Users] Grandstream 100 sidetone

Chris Albertson wrote: |What firmware version do you have? program version 1.0.4.39 -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature

[Asterisk-Users] Grandstream 100 sidetone

For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssional

[Asterisk-Users] sidetone issue

I am using GS 101 and as I am new to Ip phone arena. I am finding it a bit annoying to hear sidetone, especially when both parties are talking over each other occassionally. In that case, I cannot hear the other party's conversation. Is there any way to suppress it? Is it only GS or it applies

[Asterisk-Users] Grandstream 10

Resolved my problem: somehow, the setting of sip.conf has to be in this order [general] disallow=all allow=alaw allow=ulaw allow=gsm The codec in use is alaw. It works but I would like to use gsm if possible. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptogr

[Asterisk-Users] Grandstream 101

Just got GS 101 phone and plugged into the network. Got ip setup however, the following problems arise: 1. when dialing an extension, I cannot further send any key tone to Asterisk. 2. there is no sound coming from the other end. I have a sip.conf setup for GS: [General] disallow=all allow=ulaw

[Asterisk-Users] Zap show channel

What are the meaning of these Zap show channel output? Caller ID string: Owner: Real: Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax handled: no Pulse phone: no Echo Cancellation: 0 taps unles

[Asterisk-Users] Brandwidth for making internet calls

My ADSL connection speed is 512Kb up and 128Kb down. When making calls from Asterisk to IAX and back to the Asterisk, the sound is choppy and 20% of voice messages was lost. What is the production bandwidth requirement per internet call. I understand there is no guarantee of QoS but at least a

[Asterisk-Users] echo cancellation

echo cancellation is activated in /etc/asterisk/zapata.conf However, how to confirm it? Does "zap show channel 1" confirm the existence of echo cancellation? -- David Kwok Iaxtel/FWD # 17001813482 smime.p7s Description: S/MIME Cryptographic Signature

[Asterisk-Users] words for Alison

Call forwarding Call forwarding not reply call forwarding busy David Kwok smime.p7s Description: S/MIME Cryptographic Signature

[Asterisk-Users] hardware requirements - asterisk

In relation to voice degradation when having 2 or more connection to Asterisk. The comment on the network setup is quite possible. I am not too familiar with linux. How do I check whether the asterisk server's nic is running at full-duplex mode. Does Asterisk use the sound card on the box to d

[Asterisk-Users] hardware requirements of asterisk

I have been playing with 2 Asterisk boxes for testing purposes, it has been going very well. The 2 boxes are PII celeron 400 (HP Deskpro) with sound cards and lan. I have iax connecting the 2 boxes. For making cals and testing out recorded message for 1 connection it was working quite well. Ho

[Asterisk-Users] cisco 7910 phone

Hi All Will cisco 7910 ip phone compatible with Asterisk? I know that 7960 are fine. David Kwok smime.p7s Description: S/MIME Cryptographic Signature

[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2413 - 13 msgs

--__--__-- Message: 1 From: Terence Parker <[EMAIL PROTECTED]> Date: Fri, 9 Jan 2004 11:25:23 +0800 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Problem registering FWD Reply-To: [EMAIL PROTECTED] --Apple-Mail-1-822243116 Content-Transfer-Encoding: 7bit Content-Type: text/plain; charset

[Asterisk-Users] [Fwd: reject connect from iaxtel.com]

. [general] register: dkwok:[EMAIL PROTECTED] But you have to set up a client to allow Iaxtel to put calls through. [iaxtel] type=friend host=iaxtel.com context=from-iaxtel For type, type=friend * will allow both incoming and outgoing type=user * will allow incoming type=peers * will allow outgoing