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...DI/71-1' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/test-b6369010' status is 'CHANUNAVAIL' == Using SIP RTP CoS mark 5 -- Executing [9819213 at from_test:1] Set("SIP/test-09f23d18", "CALLERID(name)=James Shigley") in new stack -- Executing [9819213 at from_test:2] Set("SIP/test-09f23d18", "CALLERID(number)=4099819213") in new stack -- Executing [9819213 at from_test:3] Set("SIP/test-09f23d18", "CALLERID(all)=James Shigley<4099819213>") in new sta...

It might be worth mentioning the voip call is coming from a number we have thru bandwidth.com in case anyone uses them. James Shigley From: James A. Shigley Sent: Wednesday, December 02, 2009 3:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Variable Name needed That wasn't it either. I tried a few other likely fields from that page none of which gave the corr...

...side the building calls. So where do I check to see if it is or isn't attaching caller ID. Let's assume I'm right and the CID is the issue; What config and/or context do I need to change so that the when a queue tries to place a call to an agent there is caller ID? James Shigley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100719/97c31736/attachment-0001.htm

...(STDOUT, "STREAM FILE /var/lib/asterisk/sounds/1 ###\n"); fwrite(STDOUT, "wait for digit 10\n"); $Q1 = fgets(STDIN); // This section is where the email would be generated, but since that section works. No point making this any longer than need be. ?> James Shigley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090504/92fe4d3e/attachment.htm

...ime (1:0/0/1) -- Auto fallthrough, channel 'DAHDI/52-1' status is 'CHANUNAVAIL' -- Hungup 'DAHDI/52-1' Now all of my lines are NOT ties up so there is available paths for the call to go out. Anyway so how would I accomplish this transfer of sorts? James Shigley Monroe Telephone Answering Service 409-981-9750 Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information =is inte...

...inf,${EXTEN},1) ${ddwhome}=SIP/ddwhome Now the odd thing is when it gets the Error 14 message then the third step to dial_inf does not execute. Though when it rarely does connect with the sip phone if no one answers in 21 seconds than it will roll over to that step. Any ideas? James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information =is inte...

...;From: $from"; $message ="$UNIQUEID , $CALLERID , $EXTEN , $attachment"; mail($to,$subject,$message,$headers); ?> So is it anything obviously wrong with the script I'm missing? Besides something not being configured in php.ini correctly any other ideas? James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information is intende...

...side the building calls. So where do I check to see if it is or isn't attaching caller ID. Let's assume I'm right and the CID is the issue; What config and/or context do I need to change so that the when a queue tries to place a call to an agent there is caller ID? James Shigley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100716/c2bb75ff/attachment-0001.htm

...unt and trying to avoid an AGI script if possible I'm not sure how to email the recording(s). I also want to be able to structure the body of the email so that it reads something like You have a new call from $CallerID - "$CallerName" on 'DateTime' ... ect, ect. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information is intende...

...SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:+14092933193@> Content-Length: 0 <------------> Thank You for your time, and I apologize if this is a repeat question. I did Google, and search thru my * email archive (back thru April 09) for an answer first. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information =is inte...

...et_config: Ignoring bindaddr on reload [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring bindaddr on reload [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring bindaddr on reload == Loaded firmware 'iaxy.bin' egg*CLI> Any ideas? James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information =is inte...

...o an interface/context. We are an answering service. If no agent is available in the queue I want to send it to the Interface which goes to my TAS equipment. So DAHDI/g2/Exten How do I accomplish that because I can't figure it out from googling or http://www.voip-info.org/ James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information =is inten...

AGI is executable. =20 Then run 'agi debug' from the asterisk cli, place a call and see what was send and receive from your agi =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A. Shigley Sent: April-23-09 12:26 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] AGI PHP script =20 I have the below script that doesn't seem to be working. I don't know if I have something in the script wrong that I am just missing. Or if I don't have the php.ini set corre...

is executable. Then run 'agi debug' from the asterisk cli, place a call and see what was send and receive from your agi From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A. Shigley Sent: April-23-09 12:26 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] AGI PHP script I have the below script that doesn't seem to be working. I don't know if I have something in the script wrong that I am just missing. Or if I don't have the php.ini set correct...

Telco, location, ect?) At X times of day? =20 Ect, ect. =20 It sounds like bleed over, which can be causes by some many things the best place to start is to find a pattern if there is one. =20 James Shigley Monroe Telephone Answering Service =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David @ULC Sent: Tuesday, May 19, 2009 1:17 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Ghost ?? =20 We are using ast...

...TINUE) exten => t,1,Set(MACRO_RESULT=CONTINUE) exten => talk,1,NoOp(Caller accepted) [Inbound] exten => 4095551212,1,NoOP() exten => 4095551212,n,Dial(LOCAL/99991112223333&LOCAL/99992223334444,40) exten => 4095551212,n,Voicemail(1002 at default) James Shigley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100224/dd81102a/attachment.htm

Hi I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things I do on INVITES is to re-authenticate the user from OpenSER. Then when the INVITE gets passed to Asterisk I capture the AUTH to a variable in the dialplan and pass to an AGI script. I am now trying to set the same thing up in 1.6 However because the argument delimter in 1.6 has changed from pipe to comma this breaks as the

Hello All - I've been poking around the past few weeks, trying to familiarize myself with all of this. I am new to Linux, VoIP and Asterisk -- to be complete. This is my first exposure to all of these technologies. I installed AsteriskNow on my old dual Pentium 833mhz Dell PowerEdge 2400 and the install went well. I can log in and poke around in Linux and I even configured the box to be

Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [root at asterisk asterisk]# cat