similar to: Conditional "includes" in iax.conf

Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

Hi all- It's often mentioned here that recompiling the Linux kernel may improve the performance of asterisk. My question is: Does re-compiling the kernel have any effect if there are no changes to the configuration? What parameters will typically improve performance? Thanks for any insight! Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London,

Is anyone using asterisk with fail2ban? I have it working except it takes way more break-in attempts than what is set in "maxretry" in jail.conf For example, I get an email saying: "The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts against ASTERISK." when "maxretry = 5" in jail.conf Perhaps someone else is experiencing this or has resolved it,

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

Hello, I run a small network consulting company in the Chicago area and I have a client who is interested in doing an asterisk based VOIP installation. My company does not have the necessary experience to carry out the project alone so I am looking for an asterisk guru to lead the project. I'm interested in someone from the Chicago or northwest Indiana area who is very experienced with

Good morning, I am trying to use the Windows iax client. My iax.conf looks like this: [general] port=5036 bindaddr=10.1.3.111 bandwidth=high allow=gsm ; Always allow GSM, it's cool :) tos=lowdelay [pos| type=friend context=default auth=plaintext secret=pos deny=0.0.0.0/0.0.0.0 permit=10.1.3.0/255.255.255.0 host=dynamic defaultip=10.1.3.2 In the registrations dialog

Please can someone help me with the following two error messages: Error 1. I have loaded the Zaptel dirvers and everything is ok with ztcfg. I have configured Zapata.conf and everthing looks good but it apears the Zap channels dont load when starting Asterisk. When I make a call to one of the fxs port I get the following error message. -- Executing Dial("SIP/39-b204",

Hello- Is it possible to set the CDR record field called "accountcode" from within the dialplan? Or is there another way to cause this field to be set, preferably without using AGI code. Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England www.evtmedia.com

I have several of incoming numbers on IAX from voiptalk and magrathea but have a problem with IAX.conf. If I follow the example from voiptalk [VoIPTalk Incoming Number] type=friend username=VoIPTalk Incoming Number context=[XXXXXXXX] and make incoming entries in IAX.conf for the numbers like below with a different entry for each number pointing to a different context, incoming numbers always

When loading iax.conf I get warning: WARNING[884753]: chan_iax2.c:5631 set_config: Ignoring port for now In iax.conf I have: [general] port=5036 -- #Joseph

Hi I just signed up with voicepulse's voice connect service. then emailed me over configs for my extentions and iax i enter in all the info and when i start up * and do show registry it seems to be rejecting my login. Has anyone seen this before.. Any further insite will be greatly appreciated. thanks frankie (aim)cronparser (irc)crontibs 17006240093 -------------- next part

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Sorry for the stupid question: What's the purpose of defining a peer as trunk in iax.conf ? The question is also valid generally speaking (for other channel types), for instance: why define a Zap group as trunk in extension.conf ? Tnx for any help ! -- Best regards, Alessio mailto:afoc@interconnessioni.it

Running 11.13.1 on Fedora. This is a new install, but a copy of a previous - working -install. module load chan_sip Unable to load module chan_sip Command 'module load chan_sip' failed. SIP channel loading... [Oct 23 14:46:08] NOTICE[669]: chan_sip.c:31438 reload_config: Unable to load config sip.conf I don't think it's permissions: ls -ld /etc/asterisk /etc/asterisk/sip*

I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip extensions and a regular phone connected to the box. All routing works fine from the regular phone connected to the box, whether its going to FWD, broadvoice or the PSTN. The problem I am experiencing comes from making calls from the sip phones. They get routed correctly to the sip and iax trunks but when making calls

We have many IAXy devices in the field now. In all cases, in iax.conf, we have "qualify=yes", so that using "iax2 show peers", we can see whether or not the device is currently online. In some cases, the IAXy device and/or Asterisk are not communicating their qualification, because "iax2 show peers" shows the device as status UNKNOWN. However, when a user picks up

I have tried for over a month off and on to get iaxtel for inbound to work... and tonight after alot of troubleshooting we noticed this: iaxtel inbound will use the last entry in your iax.conf to auth against. So if [iaxtel] is at the top and say [voicepulse] at the bottom. An inbound call will try to auth against that [voicepulse] entry even with the [iaxtel] entry at the top of the file. Has

Hi, I am a newbie in asterisk. trying to configure firefly third party edition to connect to aserisk 1.0.3 im able to authenticate but cannot dial extensions. I have been reading the documentation cant seem to find the correct configs. Attached the error message and configs. What am I missing? *CLI> Urgent handler Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect

Hi all!, i search for some information about to setup my asterisk box with e-mail notification when a I call the voicemail application. Voicemail application works fine in the Dial Plan but nothing happens with email notification ...so what i need to know about this?...wiki pages did not help me ....thanks! G. ----- Original Message ----- From: <asterisk-users-request@lists.digium.com>

Hey all. RE: Conf Bridge. I am looking into a project that would need 8 to 10 thousand parties in a single conference. Most would be on mute but 5 to 6 would be presenters. Is the new conf bridge solid enough to handle this kind of load? Any ideas on hardware projections? If not 8 to 10 thousand how many would be realistic? If not asterisk any other suggestions. Thanks for any input.