[asterisk-users] zaptel debugging

Hi, How to enable zaptel debugging? I need to see reverse polarity messages. Thank you, Imran -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

[asterisk-users] Server Configuration for E1's

. If anybody has better solution please let me know that as well. Thank you, waiting eagerly for a response. Imran M Yousuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] (no subject)

. If anybody has better solution please let me know that as well. Thank you, waiting eagerly for a response. Imran M Yousuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] SIP Server

to the Server in SIP protocol directly?-- Imran M Yousuf ___ --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] my asterisk crashed

On 5/3/06, Goke Aruna [EMAIL PROTECTED] wrote: ... #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46 OUTBOUND_GROUP) at pbx.c:5904 #2 0xf5bbe1e4 in dial_exec_full (chan=0xa281820, data=0x0, peerflags=0xf469fee8) at app_dial.c:964

Re: [Asterisk-Users] Error : ast_readaudio_callback: Failed to write frame

On 4/30/06, Hatami Nugraha [EMAIL PROTECTED] wrote: Hi all, I always get this error message after I hangup a call, what does it mean ? WARNING[8957]: file.c:583 ast_readaudio_callback: Failed to write frame This means you hungup while asterisk was trying to play a file to you. It should be

Re: [Asterisk-Users] Speeding up the dial of DTMF's in SIP channel

. Imran On 3/15/06, Álvaro Palma [EMAIL PROTECTED] wrote: I'm dialing DTMF's in a SIP channel using the options: [sip.conf] dmtfmode=info [extensions.conf] exten = _XXX,1,Dial(SIP/gateway,,D(${EXTEN})) (this is a custom SIP gateway, which receives the DTMF's sent from softphones through

Re: [Asterisk-Users] TE411P VPM

Use: modprobe wct4xxp vpmsupport=0 On 3/1/06, Aaron Daniel [EMAIL PROTECTED] wrote: Does anyone know how to disable the VPM in software rather than removing the card altogether? The canceler isn't working as well as the software cancelers were. Aaron

Re: RE : [Asterisk-Users] lists problem, Gmail????????

compare your emails with the list archives to be sure you didnt miss something important. Also, the problems seems to have gone away this week. Regards Imran On 2/13/06, Joseph Tanner [EMAIL PROTECTED] wrote: May be some truth to it though :( Personally I use gmail, but use a different email

Re: [Asterisk-Users] echo cancel from telco

here is a little explanation: End user (You) - Your Telco -- Carrier 1 --- Carrier 2 Carrier 3 --- Carrier 4(PTT) --- Far End User So basically, the Echo cancelling work backwards usually cancellation for you would be done by

Re: [Asterisk-Users] echo cancel from telco

On 2/7/06, Imran Ahmed [EMAIL PROTECTED] wrote: here is a little explanation: End user (You) - Your Telco -- Carrier 1 --- Carrier 2 Carrier 3 --- Carrier 4(PTT) --- Far End User So basically, the Echo cancelling work

Re: [Asterisk-Users] Rtp packets being dropped

AFAIK asterisk does not drop the packets, it just turns them into silence if it detects a dtmf. On 2/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello Friends, I am experiencing a problem. The rtp packets which detect dtmf from inband are being dropped. I have tried a priority ip

Re: [Asterisk-Users] meetme and dtmf

. asterisk will never see any dtmf (which should be okay in this specific case). dtmf tones are not squelched so the other user in the conference will hear dtmf tones. Imran ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list

Re: [Asterisk-Users] meetme and dtmf

either) b) Modify meetme to broadcast dtmf to all channels in conference( All channels will work in this case). Imran ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

Re: [Asterisk-Users] meetme and dtmf

On 2/1/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Imran Ahmed wrote: Even though no IAX client supports inband dtmf, An IAX client can send inband dtmf which would have corrected your problem. No, it won't. No IAX2 client will start a DSP to listen for inband DTMF, because IAX2

Re: [Asterisk-Users] meetme and dtmf

Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. I donot know if IaxComm has inband dtmf mode available, if so enable it and see if it works. ___

Re: [Asterisk-Users] Pri Gateway Hardware

You donot need multiple asterisk boxes for a single t1. A single p4 box should be helpful, you can use digiums te110p pci card for a single pri line into the box. The same box could also be on another network dealing with SIP. On 1/9/06, Carlos Alperin [EMAIL PROTECTED] wrote: All that you

Re: [Asterisk-Users] asterisk 1.2.1 and mixmonitor problem

I think the broken pipe issue is related with the mpg123 player, try disabling moh and see if it behaves the same way On 12/19/05, Maximiliano J. Goldsmid [EMAIL PROTECTED] wrote: I have the same problem !! :-( 2005/12/18, Mohammad Shokuie [EMAIL PROTECTED]: Hi there, Any one

[Asterisk-Users] number of users in a meetme conference

Hi All, I want to know what is the maximum number of users allowed in a single meetme conference. How far is this number practically feasible Thanks Imran ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list

[Asterisk-Users] Incoming context problem

.. i didn`t had to do anything.. i want to use asterisk to attend the call and forward to a extension. how shld write the context for sip and extension.conf ? best wishes Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] Incoming context problem

.. i didn`t had to do anything.. i want to use asterisk to attend the call and forward to a extension. how shld write the context for sip and extension.conf ? best wishes Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] Sip calling errors

allow=g723.1 allow=alaw allow=ulaw allow=gsm allow=g729 [venus] type=friend context=sip-dial host=2.2.2.2 canreinvite=no qualify=yes disallow=all allow=gsm allow=g729 insecure=very dtmfmode=info tos=0x18 [] type=friend host=dynamic username= secret=imran dtmf=inband context=internal

[Asterisk-Users] Fwd: Sip calling errors

-- Forwarded message -- From: iMRAN [EMAIL PROTECTED] Date: May 1, 2005 12:16 PM Subject: Sip calling errors To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Alexander Scheerschmidt [EMAIL PROTECTED] Hi Pros, I`m new to Asterisk

[Asterisk-Users] SIP Errors from MP108 please help - confs included

allow=g723.1 allow=alaw allow=ulaw allow=gsm allow=g729 [venus] type=friend context=sip-dial host=2.2.2.2 canreinvite=no qualify=yes disallow=all allow=gsm allow=g729 insecure=very dtmfmode=info tos=0x18 [] type=friend host=dynamic username= secret=imran dtmf=inband context=internal dtmfmode

[Asterisk-Users] SIP calling Error from MP108 please help - confs included

=no disallow=all allow=g723.1 allow=alaw allow=ulaw allow=gsm allow=g729 [venus] type=friend context=sip-dial host=2.2.2.2 canreinvite=no qualify=yes disallow=all allow=gsm allow=g729 insecure=very dtmfmode=info tos=0x18 [] type=friend host=dynamic username= secret=imran dtmf=inband context=internal

[Asterisk-Users] SIP calling Error from MP108 please help - confs included

allow=g723.1 allow=alaw allow=ulaw allow=gsm allow=g729 [venus] type=friend context=sip-dial host=2.2.2.2 canreinvite=no qualify=yes disallow=all allow=gsm allow=g729 insecure=very dtmfmode=info tos=0x18 [] type=friend host=dynamic username= secret=imran dtmf=inband context=internal

[Asterisk-Users] Route SIP calls to provider

thankyou all for reading this mail and i hope someone will be kind enough to help. Best regards, Imran ___ 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] Newbie - VoIP route SIP calls to provider

thankyou all for reading this mail and i hope someone will be kind enough to help. Best regards, Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

[Asterisk-Users] Newbie - VoIP route SIP calls to provider

thankyou all for reading this mail and i hope someone will be kind enough to help. Best regards, Imran ___ 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] VoIP route SIP calls to provider

thankyou all for reading this mail and i hope someone will be kind enough to help. Best regards, Imran ___ 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] Want to use Asterisk instead of existing Meridian Norstar system ... need some help

phones) How should the extension jacks in the offices be wired I guess this will depend on the answer to # 2 above. Any help will be greatly appreciated. Best regards Imran -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus

Re: [Asterisk-Users] SIP Softphone for testing with Asterisk

Hi Ray I`m using SJphone softphones with my * , working fine.. Imran On Apr 7, 2005 8:41 AM, raymond [EMAIL PROTECTED] wrote: Hi all, I had just set up my asterisk server. Can anybody know that is there any sip softphone for testing with asterisk? (I had download some from internet

Re: [Asterisk-Users] fedora 3

Hi, I`ve installed on FC-3 last month and its working gr8... no probs so far Imran On Apr 6, 2005 2:38 PM, Altus Snyman [EMAIL PROTECTED] wrote: Good day all I have a Fedora core 3 installation Is there any hassles with asterisk? Thanks Altus

[Asterisk-Users] Audio codec MP108 please help

hi all, can any 1 pls tell me the context i shld add on sip.conf for Audiocodec MP108 8 fxs please. i want to add 2 phone on MP108 port assign extention and dial each other, can`t get a dialtone only busy signal. Thnx ppls Imran ___ Asterisk-Users

[Asterisk-Users] What is ZAP ? newbie question sorry

Hi Pros, Please advice whats the purpose of ZAP, if i have softphones and ATA 186 with PSTN trunk, wht ZAP will do ? do i zap to route calls internal softphone to softphones ? thnx a lot Ronny ___ Asterisk-Users mailing list

[Asterisk-Users] Audio codec MP108

hi all, can any 1 pls tell me the context i shld add on sip.conf for Audiocodec MP108 8 fxs please. can`t get a dialtone only busy signal. Thnx ppls Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com

[Asterisk-Users] lost newbie requesting help for Asterisk Implementation

Hi friends, i`m totally a newbie on VoIP let alone asterisk. I`m very much interested in learning asterisk to deploy on my small Call Center, we have 2 audioCodec MP-108 8 fxs port SIP device and 6 A800 H323 analog quintums. I installed fresh asterisk with samples, might b peice of cake for u

[Asterisk-Users] SIP H323 gateway

could use Asterisk for this ? My question what files I need to modify on asterisk for SIP and H323 working and place calls. Can any 1 send a sample config for SIP and h323 please? Best regards, Mohd. Imran Kamal ___ Asterisk-Users mailing list Asterisk-Users

[Asterisk-Users] What is the best and easiest flavor to be used with Asterisk.

have to hard reset the machine to bring it back up. Any suggestions will be greatly appreciated. Thanks Imran Sadiq Systems Engineer Tel: +64 9 377 8282 World Class Support for any business Fax: +64 9 377 7900

[Asterisk-Users] Line drops after 5-10 seconds

, Imran Sadiq Systems Engineer Tel: +64 9 377 8282 World Class Support for any business Fax: +64 9 377 7900 with between 7 and 70 computers. Mob: 027 286 9269 LANcom

RE: [Asterisk-Users] Just saw your [Asterisk] xJack Segfault in Asterisk

Well, I had to compile the Mepis source and install it again. It did compile the Zaptel drivers but then is started giving other problems. Like it would not instalel ztdummy. Therefore I have given up on Mepis and downloading Red Hat 8.0 now. I will install Asterisk on that now. Thanks for

[Asterisk-Users] Problems installing Zaptel

Hi, I am new to asterisk. I have downloaded Asterisk and Zaptel from the cvs root. I am installing them on Mepis with linux-2.6.7 Whenever I try to do make in the zaptel directory, I get the following errors. make C /lib/modules/`uname r` /build SUBDIRS=/usr/src/zaptel modules

RE: [Asterisk-Users] Problems installing Zaptel

- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Wednesday, 22 December 2004 10:33 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problems installing Zaptel Imran Sadiq wrote: Hi, I am new

RE: [Asterisk-Users] Problems installing Zaptel

Fineberg Sent: Wednesday, 22 December 2004 4:56 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problems installing Zaptel Imran Sadiq wrote: Kristian,Thanks for that.It still gives me the same error.I have also tried make linux26 in the zaptel

[Asterisk-Users] digitnetworks card issues?

as I start dialing numbers I get feedback and noise, and the call doesn't go through. Any suggestions? Thanks, Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

Re: [Asterisk-Users] digitnetworks card issues?

] digitnetworks card issues? Have you contacted digitnetworks for support? This list is owned and maintained by Digium, who already gave you Asterisk for free. Probably not the best forum to ask for support for a competitive product here. -Original Message- From: Imran Akbar [mailto

[Asterisk-Users] no dial tone when dialing out on vonage

a wait period before accessing the line and exten = _XX,1,Dial(Zap/2/${EXTEN}) to no avail. what's goin on? Thanks, Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE

[Asterisk-Users] line feedback, no dial tone

Hi, after following up on my previous email about zaptel x100p having trouble accessing a vonage dial tone, I think the problem is with feedback and noise on the line - any remedies for this? Thanks, Imran ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] international caller id support

. thanks Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] transferring call to another line

for ringing that line (fxs)? thanks Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] extensions = s,1,Dial(Zap/2/number) noise

Hi, I'm trying to answer a call on one line and dial out a number on a zaptel x100p fxo, but all I get from the phone I'm dialing is silence after it is picked up, and on the line that's supposed to be dialed out itself, noise. Thanks, Imran

Re: [Asterisk-Users] Re: zaptel configuration

Asalamualaikum Atif, i saw your guy's ad in spider magazine. sounds cool... yeah, i got asterisk to work, i had to build zaptel before asterisk. just trying to transfer from one line to another now... thanks Imran Atif Rasheed wrote: well Imran, I am not a guru of Asterisk, but I think

[Asterisk-Users] Dial/Zap doesn't work

to select. thanks Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] zaptel configuration

they mentioned this is because some chan_zap.so file isn't loaded because of the zaptel installation. I was told I had to REINSTALL asterisk after the zaptel stuff, which again didn't do anything. How can this be so hard to even get installed? Thanks, Imran

Re: [Asterisk-Users] zap show channels - no such command

for two FXO cards. Thanks el Flynn wrote: Imran Akbar wrote: edited the zaptel.conf, zapata.conf, extensions.conf to proper settings. added chan_zap.so to modules.conf, when asterisk starts up it can't find it. Why don't you post a snippet of the zaptel stuff as reported by dmesg? That may help

Re: [Asterisk-Users] zap show channels - no such command

wo, i have to rebuild asterisk after i install zaptel? where did that come from? let me try... thanks Imran Darryl Ross wrote: Hi Imran, I seem to have done the zaptel installation - what am I missing - i still don't have a chan_zap.so file? Did you rebuild Asterisk after installing Zaptel

Re: [Asterisk-Users] zap show channels - no such command

Tried recompiling asterisk after the zaptel installation... still don't have a chan_zap.so file. help anyone? Thanks, Imran Imran Akbar wrote: wo, i have to rebuild asterisk after i install zaptel? where did that come from? let me try... thanks Imran Darryl Ross wrote: Hi Imran, I seem

[Asterisk-Users] zaptel installation

so however, but it makes no difference. any hints? thanks Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

[Asterisk-Users] zap show channels - no such command

Hi, in response to a previous posting regarding getting the x100p to work, I was told to run zap show channels, but when i do i get no such command 'zap' There was a previous posting on this, but the guy never posted the solution. thanks, Imran

Re: [Asterisk-Users] zap show channels - no such command

Thanks Jeremy, but how exactly do I load chan_zap.so? I put it into my modules.conf, but when i run asterisk now it says it can't find it (loading module zap_chan.so failed). It doesn't seem to be on my system... thanks imran Jeremy McNamara wrote: Imran Akbar wrote: Hi, in response

Re: [Asterisk-Users] zap show channels - no such command

to the best of my knowledge, i have, but i'm redoing it. i'm looking at the instructions at http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation is that the best guide? thanks Imran Jeremy McNamara wrote: Imran Akbar wrote: Thanks Jeremy, but how exactly do I load chan_zap.so? I put

Re: [Asterisk-Users] zap show channels - no such command

sorry, my bad. typo in the email, but it was correct in modules.conf. Im trying to reinstall the zaptel stuff, but i'm not seeing anything in var/log/messages after doing my modprobe's? Thanks Jon Radon wrote: It should be chan_zap.so not zap_chan.so. Imran Akbar wrote: Thanks

Re: [Asterisk-Users] zap show channels - no such command

, extensions.conf to proper settings. added chan_zap.so to modules.conf, when asterisk starts up it can't find it. Thanks, Imran el Flynn wrote: Imran Akbar wrote: sorry, my bad. typo in the email, but it was correct in modules.conf. Im trying to reinstall the zaptel stuff, but i'm not seeing

[Asterisk-Users] zaptel config

,5,BackGround(demo-instruct) I have no idea why it's not working would appreciate any help Thanks, Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Local PBX

I am running Asterisk. I want to make my local PBX. I have Cisco ATA 186-I1. i want to connect two analog telephone connected to ATA 186 and make them extention to dial each other. how i can make it. Imme -- __ Sign-up for your own FREE

[Asterisk-Users] 1X1 PBX

I have Asterisk and Cisco ATA 186. How i can make small PBX. let me know the step and configuration made in conf files. Imme -- __ Sign-up for your own FREE Personalized E-mail at Mail.com http://www.mail.com/?sr=signup