[asterisk-users] Help needed for chan_ss7 for Digium device

TE420B. Can anybody provide me required ss7.conf file and also provide dahdi configuration which is needed for this device. Thanks you so much in advance!! Thanks, Max Alex -- _ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] Look c8m9kl

37pgn. http://darkskiesblog.com/wp-content/uploads/img/vosc.html 22gv7con3pr fjfvojvm e1cugvfj, tuzj2tcz2 4zssju6k5bfj. jdc52cc twdoh35sn4s sb2yj. -- Thanks, Max Alex Voip Developer -- _ -- Bandwidth and Colocation Provided

[asterisk-users] dahdi issue on digium AEX800

but i am not able to get their audio, I have disabled firewall, selinux is also off. If I am applying this line to analog phone then also it is working fine, But when it is added on digium card then this issue happens, can anybody help me for this issue? Thanks, Max Alex Voip Developer

Re: [asterisk-users] Dahdi issue on sangoma A200

Hi, Thanks for this information, but it is not working for both the issues, I have tried with the configuration with cidsignalling, cidstart etc.. Can any one provide more help for this. Thanks, Max Alex Voip Developer On Mon, Aug 9, 2010 at 5:31 PM, asteriskguru asteriskguru beaasteriskg

[asterisk-users] Dahdi issue on sangoma A200

callerid=asreceived signalling = fxs_ks channel = 3 context=from-zaptel group=0 echocancel=yes callerid=asreceived signalling = fxs_ks channel = 4 --- Please hemp me for this issues. Thanks, Max Alex Voip Developer

[asterisk-users] Moh help needed

2001 put on hold to 1001. Please let me suggestions on this. Thanks, Max Alex Voip Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

[asterisk-users] DeadAgi application issue

for this? Thanks in advance!!! Thanks, Max Alex Voip Developer ___ -- 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

Re: [asterisk-users] [asterisk-dev] Grandstream blind transfer issue

(${DIALSTATUS}) Thanks, Max Alex Voip Developer On Wed, Apr 8, 2009 at 9:47 PM, Klaus Darilion klaus.mailingli...@pernau.at wrote: Haven't you read my email? 1. Wrong list 2. Missing log entries (set debug 4, set verbose 4) klaus Max Alex schrieb: Hi All, Thanks for your reply

Re: [asterisk-users] [asterisk-dev] Grandstream blind transfer issue

' in dialplan of this. Please provide me some help. Thanks in advance!! Thanks, Max Alex Voip Developer On Wed, Apr 8, 2009 at 2:04 AM, Klaus Darilion klaus.mailingli...@pernau.at wrote: Max Alex wrote: Hi All, I have working asterisk version 1.4.24. I have a blind transfer issue with grandstream

Re: [asterisk-users] Grandstream blind transfer issue

Hi I have used the transfer operation this way. When i got a call on grandstream phone, i will receive it and press transfer button and enter transfer number and press send button. My call is disconnected but no call transfer from asterisk. Please advice me!! Thanks, Max Alex Voip Developer

[asterisk-users] Grandstream blind transfer issue

transfer button of the grandstream phone. Can anybody provide help for this issue? Thanks in advance!! Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

[asterisk-users] Asterisk crashed!!!

the cause of crash? Please checkout following link, I have uploaded coredump backtraces there. http://pastebin.com/m5480bcb8 Please provide me help regarding this. Thanks in advance. Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided

[asterisk-users] Hold/Resume issue with polycom

have put the incoming call on hold and when we try to resume it back, the call is hangup, and not able to connect the hold channel. Can anyone provide help!! Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] Need help on Forwarding

in Advance!! Thanks, Max Alex Voip Developer ___ -- 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] DeadAgi Application in asterisk 1.6

a problem with asterisk 1.6 deadagi application, when the call is hangup at that time the script is exited and no duration and status will be counted, So please provide help regarding this deadagi application in asterisk 1.6 branch, Please help me regarding this!! Thanks in Advance!!! Thanks, Max Alex Voip

[asterisk-users] EVRC support

. Please provide information!!! Thanks, Max Alex Voip Developer ___ -- 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] Realtime MOH

disappered and we must have to reload to load moh again. Can any body please help me regarding MOH configuration!! Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] Local channel Help required

. Thanks, Max Alex Voip Developer On Mon, Jan 12, 2009 at 6:45 PM, Philipp Kempgen philipp.kemp...@amooma.dewrote: Philipp Kempgen schrieb: Max Alex schrieb: If i got the NOANSWER then the channel is not passing to next priority. I need to pass that channel to the next priority

Re: [asterisk-users] Local channel Help required

Hi All, We have already use 'g' option in that, but it is not working in my case. Thanks, Max Alex Voip Developer On Sun, Jan 11, 2009 at 1:51 AM, Philipp Kempgen philipp.kemp...@amooma.dewrote: Max Alex schrieb: If i got the NOANSWER then the channel is not passing to next priority. I

[asterisk-users] Local channel Help required

) exten=s,1,Goto(${MACRO_EXTEN}|1) [macro-voicedid] exten=s,1,NoOp(${ARG1}) Please provide me help regarding this!!! Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] Asterisk CLI got freezed!!

itself or we have to setup some preventions for that. Can anybody suggest me regarding this freeze cli issue! Thanks in advance!! Thanks, Max Alex Voip Developer On Wed, Jan 7, 2009 at 7:07 PM, Grygoriy Dobrovolskyy megaho...@gmail.comwrote: 2009/1/7 Max Alex max.aster...@gmail.com Hi

Re: [asterisk-users] Asterisk CLI got freezed!!

Hi, Thanks for your reply Can you suggest me how can we avoid it by doing any configuration changes in asterisk. So the freeze issue may not be occurred again! Please provide me some help!!! Thanks in advance! Thanks, Max Alex Voip Developer On Wed, Jan 7, 2009 at 12:58 PM, Grey Man greymanv

[asterisk-users] Asterisk CLI got freezed!!

. And because of this my iaxmodems are also getting time out from asterisk. Please provide some help regarding this freeze issue. Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users

Re: [asterisk-users] Asterisk CLI got freezed!!

HI, Thanks for your reply, But we have not setup DNS servers in asterisk. Asterisk is not getting any DNS requests. Please provide help regarding this. Thanks, Max Alex Voip Developer On Tue, Jan 6, 2009 at 4:10 PM, Grey Man greymanv...@gmail.com wrote: Make sure the DNS servers Asterisk

[asterisk-users] Need help for transfer

can not allow 2102 not to forward on 2103. and also i want to prevent the SIP/2.0 302 Moved Temporarily. please advice me that how can we set the user for not to forward or transfer on 2103. i have tested with allowtransfer=no in sip. Thanks in advance! Thanks, Max Alex Voip Developer

Re: [asterisk-users] Anonymous callerid

services so our calls will not be disconnected and recieved by them. Please provide some help for this. Thanks, Max Alex Voip Developer On Sun, Nov 30, 2008 at 1:07 AM, Philipp Kempgen [EMAIL PROTECTED]wrote: Max Alex schrieb: Actully we are getting the anonymous callerid from

Re: [asterisk-users] Anonymous callerid

for this! Thanks, Max Alex Voip Developer On Fri, Nov 28, 2008 at 7:47 PM, Philipp Kempgen [EMAIL PROTECTED]wrote: Max Alex schrieb: I have one issue regarding override callerid when i have anonymous call. I have added PAI in sip header and also set sendrpid = yes in sip.conf

[asterisk-users] Disable Transfer

Hi All, I want to prevent transfer on based of user, means we can disable any user or peer to transfer calls in asterisk. Can any one helps how can we prevent transfer feature. I am using asterisk 1.4 branch. Thanks, Max Alex Voip Developer

[asterisk-users] Anonymous callerid

using asterisk 1.4 branch. thanks in advance!! Thanks, Max Alex Voip Developer ___ -- 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

[asterisk-users] Asterisk Instant message passing with eyebeam

in eye beam. and in asterisk i got Method is not implemented. Can anybody helps me in this? If any patches are there then please let me know. Thanks in advance!! Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] RTP LOG

Hi All, I am using asterisk 1.4.22 in my local system I want to know how can we set ability to log and report RTP and jitter statistics per call. Is there any configuration in logger or configuration in rtp? Please provide some guide lines for this. Thanks in advance! Thanks, Max Alex Voip

Re: [asterisk-users] RTP LOG

Hi All, Thanks for reply i have tried for this, it looks fine for me, but is there any way to check rtp log while call is connected or any way to enable it to write in log file. Please give me some guide lines! thanks in advance. Thanks, Max Alex Voip Developer On Sat, Nov 15, 2008 at 3:21 AM

[asterisk-users] SRTP support in asterisk 1.6

Hi All, I am checking srtp support in asterisk 1.6, Let me know any patches available or changes needed for srtp support in asterisk 1.6. Thanks in advance! Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] Asterisk Callerid Help Needed

: 240 User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 308 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER Supported: 100rel, x-sipura Content-Type: application/sdp can any body help me to over ride the callerid? Thanks, Max Alex Voip Developer

[asterisk-users] Asterisk custom functions

to functions and using that functions in dialplan. but it is always gives me function is not registered. can any body explain how to register custom functions in asterisk? Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Dial issue

Hi, can you please confirm that DTMF is working properly or not? Thanks, Max Alex Voip Developer On Sat, Sep 27, 2008 at 12:24 AM, equis software [EMAIL PROTECTED]wrote: Hi, when I make a call I need that the caller can** hang up by dialing ***(H option in Dial command), the call

Re: [asterisk-users] Asterisk CDR Problem for Export CSV (Asterisk-stat-v2)

Hi Hiren, Can you please confirm the php-gd is properly installed? Thanks, Max Alex Voip Developer On Tue, Sep 9, 2008 at 4:20 PM, Hiren Mistry [EMAIL PROTECTED]wrote: Dear All, I have configured here Asterisk-stat (Call Detail Records)for CDR ANALYSER. Here I am facing

[asterisk-users] Help about the Rxfax on asterisk

and suddently asterisk crashes and i can't get email notification for received faxes. any one help me about the crashes of asterisk? Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008

Re: [asterisk-users] Asterisk CDR Problem

Hi, let me know that you have configured properly in res_pgsql.conf in asterisk with proper, and it is connected properly to database with database details. Thanks, Max Alex Voip Developer On Fri, Aug 29, 2008 at 10:26 AM, Hiren Mistry [EMAIL PROTECTED] wrote: Hi , I have check zapte.conf

Re: [asterisk-users] Asterisk CLI Show Error :- (**Unknown**) instead of (Zap/22-1, )

Hi Hiren, Have you properly configured the zap channels in asterisk, which device have you configured in asterisk with zaptel? let me know the dial plan for ivr. Thanks, Max Alex Voip Developer On Thu, Aug 28, 2008 at 11:40 AM, Hiren Mistry [EMAIL PROTECTED] wrote: Hi, Everybody, I am

[asterisk-users] Voicemail has issues with DTMF

for this? -- Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update

[asterisk-users] Blind Transfer is not working in incoming calls

is also played, and dtmf is also set properly. But i am not getting why the incoming call is not transfer to any other number? Please help for this issue! -- Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api