[asterisk-users] cannot allocate memory
Hi: i have a hosted server with asterisk and a2billing as a billing plattform, when i am trying to enter the server remotely by ssh, memory error message displayed: -bash: fork: Cannot allocate memory i have 1GB RAM on the system ,and there is 15 to 25 concurrent calls on the system is'nt 1GB of RAM sufficient for this volume of calls on Asterisk. and when iam using "top" command this is what i get: top - 20:20:25 up 2:15, 1 user, load average: 0.57, 0.22, 0.13 Tasks: 100 total, 1 running, 36 sleeping, 0 stopped, 63 zombie Cpu(s): 7.7% us, 2.3% sy, 0.0% ni, 90.0% id, 0.0% wa, 0.0% hi, 0.0% si, Mem: 1048576k total, 316088k used, 732488k free, 0k buffers Swap: 0k total, 0k used, 0k free, 0k cached As i see it that the free memory is 732488k ,so it should'nt make this error. _ Windows Live™ Hotmail®…more than just e-mail. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t2_hm_justgotbetter_howitworks_022009___ -- 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] linksys PAP2t and asterisk
Hi:
yes i think this is it ,but what is it and how can i remove it ?
Date: Sat, 14 Feb 2009 14:23:27 -0700From: floj...@gmail.comto:
asterisk-us...@lists.digium.comsubject: Re: [asterisk-users] linksys PAP2t and
asteriskMan, as the CLI says:
SIP/us-092acb78 is ringing (here it gives me a fake ring)
It's the channel SIP/us/something, which is generating ring signalling.
2009/2/14 wassim Darwish
this post is attached to the prevoius post, this is what i have on CLI when i
call from Linksys pap2t to asterisk and then asterisk bridge the call to a sip
provider:-- Executing [88017736288...@direct:1] Dial("SIP/490115-092bacc8",
"SIP/us/88017736288155") in new stack-- Called us/88017736288155-- Call
on SIP/us-092acb78 left from hold-- SIP/us-092acb78 is making progress
passing it to SIP/490115-092bacc8-- SIP/us-092acb78 is ringing (here it
gives me a fake ring) how can i disable this ringing .
From: wassim...@hotmail.comto: asterisk-us...@lists.digium.comdate: Fri, 13 Feb
2009 20:08:20 +Subject: [asterisk-users] linksys PAP2t and asterisk
Hi all:when i make a call from linksys pap2t to an asterisk server a fake ring
is heard some times ,but when sending calls between 2 asterisk servers through
sip no fake ring is heard but real one. any suggestions please.
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Re: [asterisk-users] linksys PAP2t and asterisk
this post is attached to the prevoius post, this is what i have on CLI when i
call from Linksys pap2t to asterisk and then asterisk bridge the call to a sip
provider:
-- Executing [88017736288...@direct:1] Dial("SIP/490115-092bacc8",
"SIP/us/88017736288155") in new stack-- Called us/88017736288155-- Call
on SIP/us-092acb78 left from hold-- SIP/us-092acb78 is making progress
passing it to SIP/490115-092bacc8-- SIP/us-092acb78 is ringing (here it
gives me a fake ring)
how can i disable this ringing .
From: wassim...@hotmail.comto: asterisk-us...@lists.digium.comdate: Fri, 13 Feb
2009 20:08:20 +Subject: [asterisk-users] linksys PAP2t and asterisk
Hi all:when i make a call from linksys pap2t to an asterisk server a fake ring
is heard some times ,but when sending calls between 2 asterisk servers through
sip no fake ring is heard but real one. any suggestions please.
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[asterisk-users] linksys PAP2t and asterisk
Hi all: when i make a call from linksys pap2t to an asterisk server a fake ring is heard some times ,but when sending calls between 2 asterisk servers through sip no fake ring is heard but real one. any suggestions please. _ Windows Live™: E-mail. Chat. Share. Get more ways to connect. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t2_allup_explore_022009___ -- 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
[asterisk-users] fake ring again when using SIP
Hi: I cant figure it out why fake ring is heared when dialing through SIP (Asterisk) ,it often it gives me fake ring but not always ,in some calls it gives me real ring. the dialplan is without 'r' option. _ Hotmail® goes where you go. On a PC, on the Web, on your phone. http://www.windowslive-hotmail.com/learnmore/versatility.aspx#mobile?ocid=TXT_TAGHM_WL_HM_versatility_121208 ___ -- 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] fake ringback tone
Hi:
Iam not using the 'r' option in my dial plan ,here what i have in my dial plan:
[gw]exten => _70.,1,Dial,SIP/grands/${EXTEN}> Date: Fri, 9 Jan 2009 16:25:41
-0500> From: stot...@asteriskhelpdesk.com> To: asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] fake ringback tone> > On Fri, Jan 9, 2009 at 3:57
PM, wassim Darwish wrote:> > hi:> > When iam sending
calls through sip a fake ringback tone is generated and> > then call status
can't be viewed (if call is ringing,busy,offline) it just> > rings and rings.>
> Can i disable this?> >> > Thanks in advance.> >> > If you are using the r
option in your Dial statement, remove it. That> generates fake ringing. In
FreePBX, that option is under the General> settings, if plain jane Asterisk,
just remove the r in your dial line.> > -- > Thanks,> Steve Totaro>
+18887771888 (Toll Free)> +12409381212 (Cell)> +12024369784 (Skype)> >
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[asterisk-users] fake ringback tone
hi: When iam sending calls through sip a fake ringback tone is generated and then call status can't be viewed (if call is ringing,busy,offline) it just rings and rings. Can i disable this? Thanks in advance. _ Windows Live™: Keep your life in sync. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t1_allup_howitworks_012009___ -- 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
[asterisk-users] digium cards with sangoma cards
Hi: Iam an Asterisk user and i have a Sangoma A200 with 4 fxo modules and i want to buy Digium card with 4 fxo modules and insert it on the PCI besides the sangoma card ,so i will have 8 fxo channels on my asterisk box ,Is that right? Does Asterisk make errors if there is two different cards ? Thanks in advance; _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline ___ -- 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
[asterisk-users] G723 on asterisk 1.4.1
Hi: How to install and set up my asterisk server with G723 codec to send and receive calls using it. Thanks in advance; Wassim _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE ___ -- 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
[asterisk-users] polarity in zapata.conf
hi: In my zapata.conf i have 4 fxo configured channels,for fxo number 1 to 3 i added polarity reversal property but for fxo number 4 i didnt add polarity reversal property but it still giving me on cosole that fxo number 4 is polarized (because the line on fxo number 4 is not polarized). what i want to do is to not let polarity reversal take effect on fxo number 4. that what i have in my zapata.conf: answeronpolarityswitch=yes hanguponpolarityswitch=yes signalling=fxs_ks context=wassim channel => 1-3 ; signalling=fxs_ks context=wassim channel => 4 Thanks in advance; _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us ___ -- 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
[asterisk-users] FXO modules and polarity reverse
I would like to know if any body tried to connect gsm gateway with polarity reversal to fxo module at asterisk server ,and if the polarity reversal solve the problem of the answer and hangup supervison on calls .i appreciate any help.Thanks in advance;Wassim _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/___ -- 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] Realtime Mysql error
Hi: Iam using Fedora core 5 . Thanks in advance; > Date: Mon, 29 Oct 2007 10:23:28 +0530> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Realtime Mysql error>> On 10/27/07, wassim darwish> wrote:> Hi:> Iam using an asterisk server with astcc ,iam facing a problem with astcc that when the call is hangup sometimes astcc doesnt calculate the call cost and the call time and without writing the call status on cdrs table .> I tried to run this command "realtime mysql status" on the asterisk console and that what i've got:> [Oct 27 01:05:32] ERROR[2607]: res_config_mysql.c:637 mysql_reconnect: MySQL RealTime: Ping failed (2006). Trying an explicit reconnect.> Connected to [EMAIL PROTECTED], port 3306 with username root for 9 hours, 43 minutes, 39 seconds.> Can any body help with this;> Hi> what is the version of asterisk> and mysql> what distro you are using> ram _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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
[asterisk-users] Realtime Mysql error
Hi: Iam using an asterisk server with astcc ,iam facing a problem with astcc that when the call is hangup sometimes astcc doesnt calculate the call cost and the call time and without writing the call status on cdrs table . I tried to run this command "realtime mysql status" on the asterisk console and that what i've got: [Oct 27 01:05:32] ERROR[2607]: res_config_mysql.c:637 mysql_reconnect: MySQL RealTime: Ping failed (2006). Trying an explicit reconnect. Connected to [EMAIL PROTECTED], port 3306 with username root for 9 hours, 43 minutes, 39 seconds. Can any body help with this; Thanks in advance; Wassim _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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
[asterisk-users] Astcc does'nt count all calls
Hi: Iam using astcc on my asterisk server,sometimes astcc does'nt count calls by not writing them into mysql ,example: Not subtracting the call cost from the face value of the entered card number and by not writing it into cdrs table.This problem occured sometimes and not always. Is it possible that this problem occured due to launching mysql by this command : /etc/init.d/mysqld start --force Can any body help me please. Thanks in advance; wassim _ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://www.toolbar.live.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] astcc sometimes doesnt write on mysql
Hi: I noticed that astcc on my asterisk server sometimes it doesnt write on mysql ,example :when the caller hangup the call its didnt written on cdrs table nor subtract the cost of the call from the face value of caller card number.This problem occured sometimes and not always. Regards; wassim _ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://www.toolbar.live.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Musiconhold instead ringing
> From: [EMAIL PROTECTED]> To:
asterisk-users@lists.digium.com> Date: Sat, 8 Sep 2007 14:48:18 +0200> Subject:
Re: [asterisk-users] Musiconhold instead ringing>> Hi,>> Am Samstag, den
08.09.2007, 09:44 +0000 schrieb wassim darwish:>>
> From: [EMAIL PROTECTED]>>> To:
asterisk-users@lists.digium.com> Date: Fri, 7 Sep 2007 19:10:04>> -0500>
Subject: Re: [asterisk-users] Musiconhold instead ringing>> On>> Friday 07
September 2007 07:02:01 pm wassim darwish wrote:>> Hi:>>>> When i get an
incoming call, i want asterisk to make the caller hear>>>> music"musiconhold"
instead of ringing,Can any body help me with>> this?>> my guess is you'd have
to Answer() the call first, then play>> moh while> Dial()'ing the exten.>> -->
Anthony - http://messinet.com ->> http://messinet.com/~amessina/gallery> 8F89
5E72 8DF0 BCF0 10BE 9967>> 92DC 35DC B001 4A4E>>>> Hi:>> Can you represent it
in extensions Dialplan ?>> Untested, but something like this should work>>
exten => _X.,1,Answer> extem =>
_X.,2,Dial(SIP/${EXTEN}|50|m(musiconhold-class)>> You have to fix this to match
Your existing dialplan (Extensions,> SIP-Accounts...).>> It won't work without
the answer-statement.>> HTH,> Karsten>>>>
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Hi:
Thank you all for your suggestions,it worked well using Karsten suggestion.
Thanks again,
Wassim
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Re: [asterisk-users] Musiconhold instead ringing
> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Date: Fri, 7 Sep 2007 19:10:04 -0500> Subject: Re: [asterisk-users] Musiconhold instead ringing>> On Friday 07 September 2007 07:02:01 pm wassim darwish wrote:>> Hi:>> When i get an incoming call, i want asterisk to make the caller hear>> music"musiconhold" instead of ringing,Can any body help me with this?>> my guess is you'd have to Answer() the call first, then play moh while> Dial()'ing the exten.>> --> Anthony - http://messinet.com - http://messinet.com/~amessina/gallery> 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E Hi: Can you represent it in extensions Dialplan ? Regards; Wassim _ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://www.toolbar.live.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] Musiconhold instead ringing
Hi: When i get an incoming call, i want asterisk to make the caller hear music"musiconhold" instead of ringing,Can any body help me with this? Best regards; Wassim _ Get the new Windows Live Messenger! http://get.live.com/messenger/overview ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] No Dial tone came from fxs modules
> Date: Wed, 5 Sep 2007 09:21:19 -0800> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] No Dial tone came from fxs modules>> Just to be clear, I thought that dialtone provision didn't require the> power cable, just generating ring voltages? Can anyone say?>> Moj>> Anthony Messina wrote:>> On Wednesday 05 September 2007 09:09:25 am wassim darwish wrote:>>>>> Hi:>>> I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i>>> made modprobe wctdm the fxs modules is lightened but there is no dial tone>>> came from it . Can i get some help please.>>>>>>> do you have the power cable attached to it. that's what you need to generate>> a dialtone.>>>>>> >>>> ___>> --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>>> ___>> Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/>> --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 Hi: I checked the power cable and its plugged in the TDM, Is there anything else to check? _ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://www.toolbar.live.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] No Dial tone came from fxs modules
Hi: I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made modprobe wctdm the fxs modules is lightened but there is no dial tone came from it . Can i get some help please. Best Regards; Wassim _ Windows Live Spaces is here! It’s easy to create your own personal Web site. http://spaces.live.com/signup.aspx ___ --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] Zaptel modules are being installed in different directory
> Date: Sun, 2 Sep 2007 04:31:22 -0500> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Zaptel modules are being installed in different directory>> Tzafrir Cohen wrote:>> On Sat, Sep 01, 2007 at 01:23:47PM -0500, Eric ManxPower Wieling wrote:>>> Tzafrir Cohen wrote:>>>> On Sat, Sep 01, 2007 at 07:30:37AM +, wassim darwish wrote:>>>>> Hi:>>>>> Iam running kernel is 2.6.8.1-12mdk but the modules of zaptel are>>>>> being installed to /lib/modules/2.6.8.1-12mdkcustom>>>>> how can i fix this up, any one have an idea?>>>> What is the output of:>>>>>>>> rpm -qa | grep kernel>>>> ls -l /lib/modules/`uname -r`/build/.config>>>>>>>>>> The easiest thing to do is remove the 2.6.8.1-12mdkcustom directory,>>> create a symlink from>>> 2.6.8.1-12mdkcustom to 2.6.8.1-12mdk, and reinstall zaptel.>>>> Are you sure?>>>> I suspect a:>>>> modinfo ./zaptel.ko>>>> would show that the "kernel version" string there is 2.6.8.1-12mdkcustom>> and hence it won't load.>>>> I suspect that Steve's advice is generally correct. I just wonder if>> there's a more "striahgt-forward" way on Mandrake, or if they give a>> different kernel version on purpuse in case you decide to rebuild the>> kernel.>>>> You have not replied to my other questions (what kernel source / headers>> rpm packages are installed). Anybody here more familiar with Mandriva's>> current magics on this subject?>>>>> You edit /usr/src/linux/Makefile and modify the EXTRAVERISON setting.> Then you spend a while trying to get the zaptel build process to not> keep installing the modules in the wrong directory, get frustrated, give> up, rm -f the zaptel source, untar zaptel, build install. Then wish you> had just done the symlink in the first place.>>>> ___> --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 Hi: Thank you all for your suggestions,i tried hard at mandrake but without any new results. so i tried to install Fedora core 5 ,Every thing went well ,i did modprobe wctdm and the TDM activated properly but when i tried to open /etc/zaptel.conf to configure i didnt find it,There is no zaptel.conf in /etc . I appreciate your help Thanks; Wassim _ Windows Live Spaces is here! It’s easy to create your own personal Web site. http://spaces.live.com/signup.aspx ___ --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] Zaptel modules are being installed in different directory
> Date: Sat, 1 Sep 2007 16:03:46 -0400> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Zaptel modules are being installed in different directory>> wassim darwish wrote:>> > Date: Sat, 1 Sep 2007 13:23:47 -0500> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Zaptel modules are being installed in different directory>> Tzafrir Cohen wrote:>> On Sat, Sep 01, 2007 at 07:30:37AM +, wassim darwish wrote:>>> Hi:>>> Iam running kernel is 2.6.8.1-12mdk but the modules of zaptel are>>> being installed to /lib/modules/2.6.8.1-12mdkcustom>>> how can i fix this up, any one have an idea?>>>> What is the output of:>>>> rpm -qa | grep kernel>> ls -l /lib/modules/`uname -r`/build/.config>>>>> The easiest thing to do is remove the 2.6.8.1-12mdkcustom directory,> create a symlink from> 2.6.8.1-12mdkcustom to 2.6.8.1-12mdk, and reinstall zaptel.>> This is a common issue for Mandrake/Mandriva users. I'm sure there is a> more correct fix, but I do the symlink thing and never have to worry> about it.>> ___> --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>>>> Hi:>> Iam new to linux world, Can you just explain how can i create symlink from 2.6.8.1-12mdkcustom to 2.6.8.1-12mdk ?>>>>> I think the more correct fix is to edit the name of the kernel and> remove the "custom" before compiling it.>> I rename all of my custom kernels but never played with this in Mandrake.>> Thanks,> Steve>>>> ___> --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 Hi: i have in my /lib/modules before installing zaptel-1.4.5 this: [EMAIL PROTECTED] modules]# ls 2.6.8.1-12mdk/ but after installing zaptel-1.4.5 this : [EMAIL PROTECTED] modules]# ls 2.6.8.1-12mdk/ 2.6.8.1-12mdkcustom/ By the way when i have done installing zaptel-1.4.5 i tried to get to /etc/zaptel.conf but it seems not exists. _ Windows Live Spaces is here! It’s easy to create your own personal Web site. http://spaces.live.com/signup.aspx ___ --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] Zaptel modules are being installed in different directory
> Date: Sat, 1 Sep 2007 13:23:47 -0500> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Zaptel modules are being installed in different directory>> Tzafrir Cohen wrote:>> On Sat, Sep 01, 2007 at 07:30:37AM +, wassim darwish wrote:>>> Hi:>>> Iam running kernel is 2.6.8.1-12mdk but the modules of zaptel are>>> being installed to /lib/modules/2.6.8.1-12mdkcustom>>> how can i fix this up, any one have an idea?>>>> What is the output of:>>>> rpm -qa | grep kernel>> ls -l /lib/modules/`uname -r`/build/.config>>>>> The easiest thing to do is remove the 2.6.8.1-12mdkcustom directory,> create a symlink from> 2.6.8.1-12mdkcustom to 2.6.8.1-12mdk, and reinstall zaptel.>> This is a common issue for Mandrake/Mandriva users. I'm sure there is a> more correct fix, but I do the symlink thing and never have to worry> about it.>> ___> --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 Hi: Iam new to linux world, Can you just explain how can i create symlink from 2.6.8.1-12mdkcustom to 2.6.8.1-12mdk ? _ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://www.toolbar.live.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] Zaptel modules are being installed in different directory
> Date: Sat, 1 Sep 2007 11:55:28 +0300> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Zaptel modules are being installed in different directory>> On Sat, Sep 01, 2007 at 07:30:37AM +0000, wassim darwish wrote:>>>> Hi:>> Iam running kernel is 2.6.8.1-12mdk but the modules of zaptel are>> being installed to /lib/modules/2.6.8.1-12mdkcustom>> how can i fix this up, any one have an idea?>> What is the output of:>> rpm -qa | grep kernel> ls -l /lib/modules/`uname -r`/build/.config>> --> Tzafrir Cohen> icq#16849755 jabber:[EMAIL PROTECTED]> +972-50-7952406 mailto:[EMAIL PROTECTED]> http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir>> ___> --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 Hi : The output of "uname -r" is "2.6.8.1-12mdk" The output of "ls -l /lib/modules/2.6.8.1-12mdk/build/.config" is "-rw-r--r-- 1 root root 60428 Oct 1 2004 /lib/modules/2.6.8.1-12mdk/build/.config" _ Call friends with PC-to-PC calling -- FREE http://get.live.com/messenger/overview ___ --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
[asterisk-users] Zaptel modules are being installed in different directory
Hi: Iam running kernel is 2.6.8.1-12mdk but the modules of zaptel are being installed to /lib/modules/2.6.8.1-12mdkcustom how can i fix this up, any one have an idea? Best Regards; Wissam _ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://www.toolbar.live.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
[Asterisk-Users] callback script
Hi: Once i have seen the post of Darren Wiebe of suggestion of a callback configuration in extensions.conf and it was like this: [callback] exten => _.,1,AGI(callback.agi,LAKEVIEW,1234567890,9998,,meetme,enhanced-outgoing) But i didnt know what to add in meetme and enhanced-outgoing contexts. if any body knows about this configuration ,just show me what to put in the meetme and enhanced-outgoing contexts and what to edit in this part of callback.agi script: $outgoingclid = ""; $channel = ""; $context = ""; $church = ""; Regards; wassim __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com ___ --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] where can I find this property: answeronpolarityswitch
where can i find these properties: answeronpolarityswitch and hanguponpolarityswitch. Regard; wassim __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] sip problem
i have configured a sip phone to make calls through a sip server but when i make call through the sip phone to the sip server every thing goes well and the call is done perfectly but on sip server it gives me these messages(i have 2 pc with different ips one with a sip phone and the another with an asterisk ): Aug 12 18:19:11 NOTICE[12149]: chan_sip.c:5284 register_verify: Peer 'wassim' is trying to register, but not configured as host=dynamic Aug 12 18:19:11 NOTICE[12149]: chan_sip.c:8730 handle_request_register: Registration from '' failed for '195.112.214.98' Aug 12 18:19:11 WARNING[12149]: chan_sip.c:7910 handle_response: Forbidden - wrong password on authentication for REGISTER for 'wassim' to '195.112.214.98' Aug 12 18:19:31 NOTICE[12149]: chan_sip.c:4380 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again any body have an idea. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] iax to iax severs
if any one can tell me how to configure the iax files of 2 iax servers ,one behind the NAT and the other real ip and the one behind the NAT requesting the other with real ip. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] most stable linux to build business
what is the most stable linux that we can build business on it, i mean the best linux a linux without problems . __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] cannot find channel_pvt.h
when i tried to compile asterisk-oh323 i get an error that channel_pvt.h is missing,where i can find and download it and in which directory i must put it. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] how to compile asterisk-oh323
if any one can tell how to compile asterisk-oh323 and what it is dependencies. Regards; wassim Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ 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] problems with compiling asterisk-oh323
i ve downloaded
asterisk-oh323-0.6.6.tar.gz
I am getting this and anybody know howto fix this?
#tar zxvf asterisk-oh323-0.6.6.tar.gz
oh323]# cd asterisk-oh323-0.6.6
asterisk-oh323-0.6.6]# ls
asterisk-driver CONFIGURATION Makefile rpm
TESTS
BUGS COPYINGREADMErules.mak
wrapper
asterisk-oh323-0.6.6]# make
for x in wrapper asterisk-driver; do make -C $x build
|| exit 1 ; done
make[1]: Entering directory
`/home/wassim/asterisk-oh323-0.6.6/wrapper'
./check_ver /root/src/oh323/pwlib pwlib
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
cat: /root/src/oh323/pwlib/version.h: No such file or
directory
cat: /root/src/oh323/pwlib/version.h: No such file or
directory
cat: /root/src/oh323/pwlib/version.h: No such file or
directory
./check_ver /root/src/oh323/openh323 openh323
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
cat: /root/src/oh323/openh323/version.h: No such file
or directory
cat: /root/src/oh323/openh323/version.h: No such file
or directory
cat: /root/src/oh323/openh323/version.h: No such file
or directory
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
g++ -Wall -x c++ -Os -DUSE_OLD_CAPABILITIES_API=1
-DWRAPTRACING -DWRAPTRACING_LEVEL=5
-DPWLIBVERSION=\"..\" -DOPENH323VERSION=\"..\"
-I/root/src/oh323/pwlib/include
-I/root/src/oh323/openh323/include
-I/root/src/oh323/openh323/include/openh323
-I../asterisk-driver -c wrapper_misc.cxx -o
wrapper_misc.o
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
In file included from wrapper_misc.cxx:34:
wrapper_misc.hxx:35:19: ptlib.h: No such file or
directory
In file included from wrapper_misc.cxx:34:
wrapper_misc.hxx:61: error: expected class-name before
'{' token
wrapper_misc.hxx:63: error: `PMutex' has not been
declared
wrapper_misc.hxx:63: error: ISO C++ forbids
declaration of `PCLASSINFO' with no type
wrapper_misc.hxx:63: error: ISO C++ forbids
declaration of `parameter' with no type
wrapper_misc.hxx:68: error: `BOOL' does not name a
type
wrapper_misc.hxx:73: error: `PString' does not name a
type
wrapper_misc.cxx: In constructor
`WrapMutex::WrapMutex(char*)':
wrapper_misc.cxx:48: error: class `WrapMutex' does not
have any field named `PMutex'
wrapper_misc.cxx:50: error: `name' undeclared (first
use this function)
wrapper_misc.cxx:50: error: (Each undeclared
identifier is reported only once for each function it
appears in.)
wrapper_misc.cxx:50: error: `PString' undeclared
(first use this function)
wrapper_misc.cxx:51: error: `cout' undeclared (first
use this function)
wrapper_misc.cxx:51: error: 'class WrapMutex' has no
member named 'Class'
wrapper_misc.cxx:51: error: `endl' undeclared (first
use this function)
wrapper_misc.cxx: At global scope:
wrapper_misc.cxx:54: error: `BOOL' does not name a
type
wrapper_misc.cxx: In member function `void
WrapMutex::Signal(const char*, int, const char*)':
wrapper_misc.cxx:78: error: `PMutex' has not been
declared
wrapper_misc.cxx:78: error: no matching function for
call to `WrapMutex::Signal()'
wrapper_misc.cxx:77: note: candidates are: void
WrapMutex::Signal(const char*, int, const char*)
wrapper_misc.cxx:79: error: `cout' undeclared (first
use this function)
wrapper_misc.cxx:79: error: 'class WrapMutex' has no
member named 'Class'
wrapper_misc.cxx:79: error: `name' undeclared (first
use this function)
wrapper_misc.cxx:79: error: `endl' undeclared (first
use this function)
make[1]: *** [wrapper_misc.o] Error 1
make[1]: Leaving directory
`/home/wassim/asterisk-oh323-0.6.6/wrapper'
make: *** [subdirs_build] Error 1
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[Asterisk-Users] does h323 exists in astcc trunks
does astcc support h323 ,because it doesnt exists h323 in trunks technology. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
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when i display g729 on iax.conf and make a call using g729 it gives this in several lines: Jul 14 17:30:58 WARNING[14196]: codec_g729.c:180 g729tolin_framein: Out of G.729 Decoder Licenses! Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ 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
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i ve tried to find a gnophone dependency " libgtksuperwin.so" i searched every where in google in wiki pages but i didnt found it at all ,if any one can help in finding it i ll be thankful,and thanks in advance. _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ 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] how to edit ring time
i dont how to edit the time for ringing "3ms" to "4ms" when it displayed on console "Nobody picked up in 3 ms" and its very short time for ringing . please if anyone can help me do it please. Sell on Yahoo! Auctions no fees. Bid on great items. http://auctions.yahoo.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] editing ring time
how to edit the time of ring "3ms" to "4ms" in astcc since it displays this on console "Nobody picked up in 3 ms" when nobody picked up the phone in 3ms and then it hangup. please help i have been asking this question from long time and no body answered me yet. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] changing "Nobody picked up in 30000 m"
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[Asterisk-Users] changing "Nobody picked up in 30000 ms"
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[Asterisk-Users] raising the sound volume on zap
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[Asterisk-Users] editing time in astcc
i dont know how to edit the time "3ms" for ringing in astcc when it says "there is no body to answer".i want to change this time to 4ms but i dont know how.please help please. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] editing time to say astcc-noanswer
i dont know how to edit the time "3ms" for ringing in astcc when it says "there is no body to answer".i want to change this time to 4ms but i dont know how.please help please. __ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html ___ 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] how does pattern routes works
i tried to write to usa destination 1* it worked well but when i tried to specify the number of digits i wrote 1NXXNXX but it did'nt work.can anybody help me please please. Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.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] problems in dialing in routes patterns
i tried to write a pattern to usa destination and that was 1* it worked well but when i tried to specify the number of digits i wrote 1NXXNXX it didnt work. so what i must to write in case to specify the number of digits of the destination. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] HOW TO WRITE ROUTE PATTERNS DIALPLAN
in routes pattern i tried to write pattern to usa destination and that was 1* it worked well but when i wanted to specify the number of digits then i tried 1NXXNXX but i didnt work.so i dont know what to write please help. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] facing troubles with routes patterns dialplan
i tried to write in routes, patterns to usa destination 1.* it worked well but i wanted to specify the number of digits then i tried 1NXXNXX but it didnt work. please help __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] failure in writing in pattern (routes)
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[Asterisk-Users] facing problems with TDM400P
i pulgged the TDM400P to my computer and when turning on and dialing on zap the following message is shown : app_dial.c:960 dial_exec_full: Unable to create channel of type 'Zap' (cause 0) == Everyone is busy/congested at this time (1:0/0/1) please help me thanks lot. __ Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html ___ 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] dialling problem with astcc
when a call comes on zap astcc.agi script launch and ask caller about his card number,and when the caller is dialing his card number(56170) sometimes astcc take it by missing a number as (5670) or doubled number as (556170) i dont know whats the problem is it from zap or is it from astcc.agi script or is it from the telephone system i dont know what to do please help. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] delay problem in asterisk
i have asterisk on my system and when making a call a delay problem in talking appears,that means when i talk to somebody he will listen me after almost a second (the ping on my voip provider's IP is 700ms to 800ms)so i dont know if the problem is in the nternet connection or another problem ,please help me i dont know what to do. __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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] HOW TO SET THE TIME TO DIAL AFTER astcc-accountnum and astcc-phonenum
when a call comes the astcc-accountnum plays and ask the caller about the card number and after playing astcc-accountnum a period of time is given for the caller to dial his card number but the problem here is the short of the time given ,and i dont know where and how can i setup the time. __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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] astcc problems
i have downloaded astcc and confiugured it on web but
the problem is when a call comes by the right callerid
it gives me on CLI like this:
-- Executing DeadAGI("Zap/1-1",
"astcc.agi|01475969|s") in new stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/astcc.agi
Detected dry run!
AGI Environment Dump:
-- accountcode =
-- callerid = 01475969
-- calleridname = unknown
-- channel = Zap/1-1
-- context = incoming
-- dnid = unknown
-- enhanced = 0.0
-- extension = s
-- language = en
-- priority = 3
-- rdnis = unknown
-- request = astcc.agi
-- type = Zap
-- uniqueid = 1112430048.4
Apr 2 03:20:54 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-tone does not exist in
any format
Res is
Silent Level is
Card no is 12345
Card has face value 3 and has used 0
3 dollars and 0 cents remain
Apr 2 03:20:54 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-youhave does not exist
in any format
-- Playing 'digits/3' (language 'en')
Apr 2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-dollars does not exist
in any format
Apr 2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-remaining does not
exist in any format
Apr 2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-badphone does not
exist in any format
-- AGI Script astcc.agi completed, returning 0
I dont know what the problem and what this warnings
mean and how can i fix them please help.
and thanks
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