similar to: Inquiry:Problem with VoiceMail

Hi, I noticed that asterisk manager interface will only accept the originate commands in sequential order. For example, if I want to ring two extensions through the AMI, and while first extension is ringing, AMI won't execute and ring second extension until first extension has answered the call. Anybody has any ideas as I had the same results even tested with telnet commands to AMI interface.

Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . Regards H.Motamedi --------------

I have an issue where a particular dialplan works but another doesn't. I'm not sure why. To me they look identical and it has me stumped. This works: [to-test] exten => _X., 1, SetCallerPres(allowed) exten => _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb) exten => _X., 3, Ringing exten => _X., 4, Dial(SIP/9330 at a-test,20,ro) exten => _X., 5,

Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my "features.conf" . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let

Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? thanks Jpc

Hello Please advice how can i configure a sip phone that is not on my local network. ie i have Xlite far some where in America and my Asterisk server is at Sahara desert . Now how can i make a call to that sip phone? Please advice what keywords to carry on?? -- Best Regards Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL:

G'day all, I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a conference. From there they should be able to bring in other people as well. This seems to be what the Asterisk n-way call HOWTO

I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. -------------- next

Dear All Can you please do me favor and let me know what are the highlights of major benefits of CentOS Release 5 (Final) over the RedHat Linux 7.2 (Enigma) as we are going to migrate to it ? Thank you in advance Regards H.Motamedi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear All It seems that our Asterisk pbx announcement files are being stored inside the "/var/lib/asterisk/sounds" folder . Can you please let us know what is the appropriate program to open and hear them on an MS Windows client ? (e.g. "pbx-invalid.gsm") Regards H.Motamedi -------------- next part -------------- An HTML attachment was scrubbed... URL:

iptables -I INPUT -s 0.0.0.0/0 -p tcp --dport 5901 -j ACCEPT I'm going strictly off memoy here so you may need to man iptables. :) hadi motamedi <motamedi24 at gmail.com> wrote: >Dear All >To open a port , I know that I need to go to "System -> Administration -> >Security Level and Firewall" -> Other ports and then I can open port-5901 as >tcp

Hi, I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: max_connection=1000 wait_timeout=60 query_cache_type=1 query_cache_limit=4M query_cache_size=512M interactive_timeout=120

Hi, I was looking round on the Internet and saw there was no definitive list of free applications available for use with Asterisk, so I thought I'd compile a list for you all. If there's anything that you know of that is actively maintained but not in the list below, let me know (bear in mind I'm not including distros or Asterisk packagings in this list). Hopefully there are a few

Hello,

Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/af1f9656/attachment.htm

Dear All Can you please let us know how to configure Asterisk to recognize extensions starting with the hash key ? Regards H.Motamedi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090801/4be38941/attachment.htm

Dear All Please be informed that I checked for the presence of internal modem on my CentOS server , as the followings : #dmesg |grep -i modem #lspci |grep -i modem #lshw |grep -i modem According to the output , it seems that my CentOS client does not contain internal modem . So I decided to add external USB modem and make use of an PCAnyWhere like application that enables for Remote PC Access .

Hi! I want to use Asterisk as load generator to test quality degradation with increased load (e.g. testing other SIP equipment or IP-links). Is anybody aware of such a setup with Asterisk - is it possible to get RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)? Thanks Klaus

Dear All Can you please let me know how can I define incoming route to accept incoming calls from an external sip server? I have defined the following profile for my sip phone : Under sip.conf : --------------------- [osaka] type=friend context=sip-outgoing host=192.168.0.139 disallow=all allow=alaw [6672019] type=friend context=sip-outgoing canreinvite=no host=dynamic nat=no Under

Dear All Can you please let us know how we can modify our Asterisk "extensions.conf" file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as "665 0000" so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one