Ryan
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Posts posted by Ryan
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Hello,
When people call in, they get the Auto Attendant. From there, they hit 1, and get sent to my Agent Group #1. From there, they may get connected with one of our agents. But the caller ID "TO:" field still shows my auto attendant's number, always, even if they have been connected to an agent.
How can I make it show that they're already out of the AA, and now connected with an agent?
Ryan
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There is no special SOAP request for that... In theory you can get the settings for the group that lists the agents and then traverse those extensions and see which one is registered. It will require some programming work, but it should be possible to do this with the current SOAP methods.
I'll probably stop posting in these forums, the answers given are just useless. No offense. Does the 'pbxnsip' user actually work on pbxnsip? Or do you just answer basic questions? You don't seem to really know the inner workings of this program nor can you efficiently answer specific questions.
I ended up figuring this out on my own, I used the XML files in the "acds" directory on the actual filesystem. This provides an XML tag called 'agents' which lists the currently available agents in that Agent Group.
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Bump...Does anyone know how I can use SOAP or XML files or any programmatic way to get who is currently available for a specific Agent Group or Hunt Group?
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It looks like the Agent Groups are more what I want rather than the Hunt Groups I'm using now. Maybe I can get what I need that way
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Well I figured I could use the GetRegistrations SOAP request for seeing who is registered; but what I really need, is whose phones are _available_. As in who is accepting calls on a certain Hunt Group or even just everyone available on the system. Obviously just being registered doesn't mean you are accepting calls.
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I would like to get a list of not only extensions that are Registered, but I would like to see who is available and who is "Do Not Disturb" or unavailable. Is there a feature that will let me see this?
Ryan
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I was able to make 1 full wav file with all my greeting parts in 1. I uploaded as the "otherwise" and now my auto attendant plays that back fine. But once it's over, it just stops, and nothing else happens. Can I get it to loop that?
EDIT: I ended up setting the timeout to 5 seconds, and now 5 seconds after my greeting it will replay, so that works fine.
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So how should I go about uploading my WAV's for the Auto Attendant to play back? I was able to get the "otherwise" to work when I uploaded a wav there, but nothing else seems to.
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Hello,
I simply want my custom WAV's to show up on the AA edit page, under Direct Destinations (like in the wiki example).
On my Auto Attendant, I go to the IVR page, type "1" in the first service flag box, I browse for my WAV file (it is 8kHz Mono and 16 bit @ 128kpbs), and click "save". It takes a sec and saves. But when I go back to Edit, I dont see the one I just uploaded under the Direct Destinations, where it says "For sales ..." etc.
Am I doing something completely wrong?
Ryan
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I am using VB so what I did was just pass the string twice but putting the admin username/password in the URL:
http://admin:pass123@mydomain.com/ajax.htm...;token=whatever
$call_index = $xml->Calls->Call->Index; // Not sure what this is
This is the call identifier which now that I know I can pull this I am planning on building a front end for our receptions to force pages to stop paging... (ie you can pass that ID to http://admin:pass123@mydomani.com/dom_call...lete_call={ID})
I was looking for a way to do this without giving the receptionist full access to the PBX system.
$call_gain = $xml->Calls->Call->Gain; // Not sure what this is either
Mine never showed anything in Gain put typically means something do to with volume amplification...
-Adam
I see. The index could be very useful then, for call barging, recording etc. I didn't even think of that.
One other thing I wanted to note, is that when getting the XML data back and converting into an array, the XML response from pbxnsip isn't in a numbered format until there is more than 1 call connected/alerting. What that means is if you have 1 connected call, the response is like:
Calls->Call->Start;
Calls->Call->From;
etc. But if there's more than 1 call coming in, it becomes numbered, like this:
Calls->Call->0->Start;
Calls->Call->0->From;
Calls->Call->1->Start;
Calls->Call->1->From;
etc etc. So if you want a foreach loop or a for loop, you can only use it if there is more than 1 call. Otherwise you simply call the data like $xml['Calls']['Call']['Start'] etc.
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I tested it here and that information may be handy... the authentication may be a problem though... in my brief testing you have to run it twice for it to work... once to login and then a 2nd time to get the information...
You are correct. But with Curl with PHP, you simply do 3 steps. Set your first curl URL to login.htm (set the login POST params), then hit the ajax.htm page, then logout.htm.
Here is the code I wrote to make this work. I hope someone finds it useful.
<?php /* * Pbxnsip Current Active Calls list * * Requires PHP 5+ because of simplexml functions * * Written by Ryan Gehrig * */ // // Set these to your server specifics // // Admin username/pass $pbx_user = 'admin'; $pbx_pass = 'pass123'; // Set server info $pbx_domain = 'http://mydomain.com'; // Main pbxnsip domain $pbx_phone_domain = 'ca.mydomain.com'; // Domain of your actual calls ################################################################################ ########################## // Ajax URL $pbx_ajax_url = $pbx_domain . '/ajax.htm?action=call_list&domain=' . $pbx_phone_domain . '&token=xxxxxxxx'; // Logout page $pbx_logout_url = $pbx_domain . '/logout.htm'; // Fields to pass for Login $postfields = "login_account=$pbx_user&login_password=$pbx_pass&login_type=auto&dom_link=dom_index.htm&usr_link=usr_index.htm"; // Setup cookies for login $cookies = 'cookies.txt'; if (!file_exists($cookies)) { $fp = fopen($cookies, 'w'); fwrite($fp, ''); fclose($fp); } // // Connect to pbxnsip interface // // Login first $ch = curl_init(); curl_setopt($ch, CURLOPT_COOKIEJAR, $cookies); curl_setopt($ch, CURLOPT_COOKIEFILE, $cookies); curl_setopt($ch, CURLOPT_URL, $pbx_domain); curl_setopt($ch, CURLOPT_POST, 1); curl_setopt($ch, CURLOPT_RETURNTRANSFER, 1); curl_setopt($ch, CURLOPT_POSTFIELDS, $postfields); curl_exec($ch); // Get Ajax page curl_setopt($ch, CURLOPT_URL, $pbx_ajax_url); $content = curl_exec ($ch); // Logout curl_setopt($ch, CURLOPT_URL, $pbx_logout_url); curl_exec ($ch); // Close up curl_close($ch); ##################################################### // // Get XML response from Pbxnsip // $xml = simplexml_load_string($content); $ret_token = $xml->Token; $ret_calls = count($xml->Calls->Call); $call_start = $xml->Calls->Call->Start; $call_from = $xml->Calls->Call->From; $call_to = $xml->Calls->Call->To; $call_state = $xml->Calls->Call->State; $call_index = $xml->Calls->Call->Index; // Not sure what this is $call_gain = $xml->Calls->Call->Gain; // Not sure what this is either $call_trunk = $xml->Calls->Call->Trunk; ?>After this you could simply echo the variables, or use some kind of XML to PHP Array function (there are a million of them) and parse it that way.
Ryan
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I got it!
I viewed the source of the "reg_calls.htm" page. It's making AJAX calls to the "ajax.htm" page. This page will return an XML response with the current calls. At least that's how it seems to be. Open the page like this (replace mydomain.com with your domain of course):
http://mydomain.com/ajax.htm?action=call_list&domain=mydomain.com&token=whatever
With no active calls, the response was:
<ResultSet> <Token>74</Token> <Calls/> </ResultSet>
When I made a phone call, I refreshed and got:
<ResultSet> <Token>73</Token> - <Calls> - <Call> <Start>2010/10/19 11:54:43</Start> <From>Ryan G (107@mydomain.com)</From> <To>xxxxxxx@mydomain.com</To> <State>connected</State> <Index>7</Index> <Gain/> <Trunk>gxw</Trunk> </Call> </Calls> </ResultSet>
And voila! (I replaced the number with xxxxxxx and domain with mydomain.com fyi).
So obviously you could just Curl request this page and anytime you need it you have a current call list.
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Right. There is this small blind spot when using AJAX! That's why multipart/mime (like MJPEG) would be better or CSTA. Anyway, what is there today is CSTA, "M-XML" might be an option later.
Alright....so, any way you can point me in the right direction? Like a specific page to query or how I can try and use this CSTA with pbxnsip? I'm going to need this as since we're switching from Asterisk (which has active call list working), and I'll need a current active call list from pbxnsip.
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Seem like using cURL could be an issue as the page is ajax-updated and lags about a second before it checks, so the page is loaded, but the background is still going.
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What is CSTA? I googled it and found a bunch of odd information. Also, have you had any users just actually use cURL to conect directly as an admin, and look at the active call list? That seems to be an option too.
Ryan
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Hello,
I would like to grab a list of current calls from pbxnsip. I don't care how I get it, even if I just have to read from an XML file or anything like that. I do have my CDR tied into MySQL, but I am not sure if the current calls get written to the database; maybe someone can clear that up.
I will be using a PHP script to parse whatever info I can find, just fyi.
Ryan
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Well I guess that address is not known with the provider (very understandable). Try to use the ANI field and the domain name in the trunk settings to put your username there.
What should I put in the ANI field? Just the phone number? What do you mean by "Try to use the ANI field and the domain name in the trunk settings to put your username there.".
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This is my dial plan. I have tried calling all kinds of variations, putting a 1 in front, 1916 in front, nothing seems to help. Should I be using that regex at the bottom? This regex used to work for other trunks before this.
100;BroadVoice;;916; 110;BroadVoice;;1916; 120;BroadVoice;;xxxxxxxxxx;1* 130;BroadVoice;;xxxxxxx;1916* 140;BroadVoice;;([0-9]*)@.*;"sip:\1@\r;user=phone"
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I'm not sure if what was provided is useful, I can't see anything useful in there other than it simply saying it's rejecting it. Is this on the broadvoice side?
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Here are any relevant (as far as I know) log entries. You can see 403 forbidden and later the 418 message, and the successful 200 registration.
[0] 2010/07/27 11:48:26: SIP Rx udp:66.245.221.192:5060: INVITE sip:7122598@sac1.cwnetpbx.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.132;branch=z9hG4bK4e6260efc060bf66 From: "Ryan" <sip:107@sac1.cwnetpbx.com>;tag=ddcee4d7b9b853b1 To: <sip:7122598@sac1.cwnetpbx.com> Contact: <sip:107@192.168.1.132> Supported: replaces Authorization: Digest username="107", realm="sac1.cwnetpbx.com", algorithm=MD5, uri="sip:7122598@sac1.cwnetpbx.com", nonce="9d407221dbdcd424532bc29f5a471e4b", response="25c5565f6ac5046fcfa44dcd946ba71a" Call-ID: 7ea9c991c0438c99@192.168.1.132 CSeq: 55344 INVITE User-Agent: Grandstream BT120 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 327 v=0 o=107 8000 8001 IN IP4 192.168.1.132 s=SIP Call c=IN IP4 192.168.1.132 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 97 9 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 [7] 2010/07/27 11:48:26: Set packet length to 20 [6] 2010/07/27 11:48:26: Sending RTP for 7ea9c991c0438c99@192.168.1.132#f831ad4403 to 192.168.1.132:5004 [0] 2010/07/27 11:48:26: SIP Tx udp:66.245.221.192:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.132;branch=z9hG4bK4e6260efc060bf66;rport=5060;received=66.245.221.192 From: "Ryan" <sip:107@sac1.cwnetpbx.com>;tag=ddcee4d7b9b853b1 To: <sip:7122598@sac1.cwnetpbx.com>;tag=f831ad4403 Call-ID: 7ea9c991c0438c99@192.168.1.132 CSeq: 55344 INVITE Content-Length: 0 [5] 2010/07/27 11:48:26: Dialplan BroadVoice: Match 7122598@sac1.cwnetpbx.com to <sip:7122598@sip.broadvoice.com;user=phone> on trunk BroadVoice [0] 2010/07/27 11:48:26: SIP Tx udp:147.135.20.221:5060: INVITE sip:7122598@sip.broadvoice.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 38.102.41.250:5060;branch=z9hG4bK-fb68b7cd1466fa636ea62cf9bd996400;rport From: "Ryan G" <sip:107@sac1.cwnetpbx.com>;tag=1038434665 To: <sip:7122598@sip.broadvoice.com;user=phone> Call-ID: b8b2ae3c@pbx CSeq: 17280 INVITE Max-Forwards: 70 Contact: <sip:9167600147@38.102.41.250:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/4.0.1.3499 P-Asserted-Identity: "9167600147" <sip:9167600147@sip.broadvoice.com> Content-Type: application/sdp Content-Length: 339 v=0 o=- 2139280778 2139280778 IN IP4 38.102.41.250 s=- c=IN IP4 38.102.41.250 t=0 0 m=audio 50212 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [7] 2010/07/27 11:48:26: Set packet length to 20 [6] 2010/07/27 11:48:26: Send codec pcmu/8000 [0] 2010/07/27 11:48:26: SIP Tx udp:66.245.221.192:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.132;branch=z9hG4bK4e6260efc060bf66;rport=5060;received=66.245.221.192 From: "Ryan" <sip:107@sac1.cwnetpbx.com>;tag=ddcee4d7b9b853b1 To: <sip:7122598@sac1.cwnetpbx.com>;tag=f831ad4403 Call-ID: 7ea9c991c0438c99@192.168.1.132 CSeq: 55344 INVITE Contact: <sip:107@38.102.41.250:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/4.0.1.3499 Content-Type: application/sdp Content-Length: 272 v=0 o=- 2035500767 2035500767 IN IP4 38.102.41.250 s=- c=IN IP4 38.102.41.250 t=0 0 m=audio 52738 RTP/AVP 0 8 9 2 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [6] 2010/07/27 11:48:26: Sending RTP for 7ea9c991c0438c99@192.168.1.132#f831ad4403 to 66.245.221.192:5004 [0] 2010/07/27 11:48:26: SIP Rx udp:147.135.20.221:5060: SIP/2.0 100 Trying Call-ID: b8b2ae3c@pbx CSeq: 17280 INVITE From: "Ryan G" <sip:107@sac1.cwnetpbx.com>;tag=1038434665 To: <sip:7122598@sip.broadvoice.com;user=phone> Via: SIP/2.0/UDP 38.102.41.250:5060;branch=z9hG4bK-fb68b7cd1466fa636ea62cf9bd996400 Content-Length: 0 [0] 2010/07/27 11:48:26: SIP Rx udp:147.135.20.221:5060: SIP/2.0 403 Forbidden Call-ID: b8b2ae3c@pbx CSeq: 17280 INVITE From: "Ryan G" <sip:107@sac1.cwnetpbx.com>;tag=1038434665 To: <sip:7122598@sip.broadvoice.com;user=phone>;tag=ijkl Via: SIP/2.0/UDP 38.102.41.250:5060;branch=z9hG4bK-fb68b7cd1466fa636ea62cf9bd996400 Allow-Events: refer User-Agent: pbxnsip-PBX/4.0.1.3499 Content-Length: 281 Content-Type: application/sdp v=0 o=644229626 2139280778 2139280778 IN IP4 38.102.41.250 s=- c=IN IP4 38.102.41.250 t=0 0 m=audio 50212 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 [7] 2010/07/27 11:48:26: Call b8b2ae3c@pbx#1038434665: Clear last INVITE [6] 2010/07/27 11:48:26: Sending RTP for b8b2ae3c@pbx#1038434665 to 38.102.41.250:50212 [0] 2010/07/27 11:48:26: SIP Tx udp:147.135.20.221:5060: ACK sip:7122598@sip.broadvoice.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 38.102.41.250:5060;branch=z9hG4bK-fb68b7cd1466fa636ea62cf9bd996400;rport From: "Ryan G" <sip:107@sac1.cwnetpbx.com>;tag=1038434665 To: <sip:7122598@sip.broadvoice.com;user=phone>;tag=ijkl Call-ID: b8b2ae3c@pbx CSeq: 17280 ACK Max-Forwards: 70 Contact: <sip:9167600147@38.102.41.250:5060;transport=udp> P-Asserted-Identity: "9167600147" <sip:9167600147@sip.broadvoice.com> Content-Length: 0 [5] 2010/07/27 11:48:26: INVITE Response 403 Forbidden: Terminate b8b2ae3c@pbx [0] 2010/07/27 11:48:26: SIP Tx udp:66.245.221.192:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.132;branch=z9hG4bK4e6260efc060bf66;rport=5060;received=66.245.221.192 From: "Ryan" <sip:107@sac1.cwnetpbx.com>;tag=ddcee4d7b9b853b1 To: <sip:7122598@sac1.cwnetpbx.com>;tag=f831ad4403 Call-ID: 7ea9c991c0438c99@192.168.1.132 CSeq: 55344 INVITE Contact: <sip:107@38.102.41.250:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/4.0.1.3499 Content-Length: 0 [0] 2010/07/27 11:48:26: SIP Rx udp:66.245.221.192:5060: ACK sip:7122598@sac1.cwnetpbx.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.132;branch=z9hG4bK4e6260efc060bf66 From: "Ryan" <sip:107@sac1.cwnetpbx.com>;tag=ddcee4d7b9b853b1 To: <sip:7122598@sac1.cwnetpbx.com>;tag=f831ad4403 Contact: <sip:107@192.168.1.132> Authorization: Digest username="107", realm="sac1.cwnetpbx.com", algorithm=MD5, uri="sip:7122598@sac1.cwnetpbx.com", nonce="9d407221dbdcd424532bc29f5a471e4b", response="0f8d30e25d489f774d114adb56d7924f" Call-ID: 7ea9c991c0438c99@192.168.1.132 CSeq: 55344 ACK User-Agent: Grandstream BT120 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 [0] 2010/07/27 11:48:26: SIP Rx udp:66.245.221.192:5060: CANCEL sip:7122598@sac1.cwnetpbx.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.132;branch=z9hG4bK4e6260efc060bf66 From: "Ryan" <sip:107@sac1.cwnetpbx.com>;tag=ddcee4d7b9b853b1 To: <sip:7122598@sac1.cwnetpbx.com>;tag=f831ad4403 Supported: replaces Call-ID: 7ea9c991c0438c99@192.168.1.132 CSeq: 55344 CANCEL User-Agent: Grandstream BT120 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 [0] 2010/07/27 11:48:26: SIP Tx udp:66.245.221.192:5060: SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.1.132;branch=z9hG4bK4e6260efc060bf66;rport=5060;received=66.245.221.192 From: "Ryan" <sip:107@sac1.cwnetpbx.com>;tag=ddcee4d7b9b853b1 To: <sip:7122598@sac1.cwnetpbx.com>;tag=f831ad4403 Call-ID: 7ea9c991c0438c99@192.168.1.132 CSeq: 55344 CANCEL Content-Length: 0 [0] 2010/07/27 11:48:30: SIP Tx udp:147.135.20.221:5060: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 38.102.41.250:5060;branch=z9hG4bK-cb476705edf7fe4d563c3ca1fdb0f244;rport From: "9167600147" <sip:9167600147@sip.broadvoice.com>;tag=2008174283 To: "9167600147" <sip:9167600147@sip.broadvoice.com> Call-ID: sdfhuypb@pbx CSeq: 16669 REGISTER Max-Forwards: 70 Contact: <sip:9167600147@38.102.41.250:5060;transport=udp;line=70efdf2e> User-Agent: pbxnsip-PBX/4.0.1.3499 Supported: outbound Authorization: Digest realm="BroadWorks",nonce="BroadWorksXgc3uzhteTn0br60BW",response="e5674182ed59e133f14504103b9ea2ce",username="9167600147",uri="sip:sip.broadvoice.com",qop="auth",nc=00004983,cnonce="48fb85e1",algorithm=MD5 Expires: 3600 Content-Length: 0 [0] 2010/07/27 11:48:30: SIP Rx udp:147.135.20.221:5060: SIP/2.0 200 OK Call-ID: sdfhuypb@pbx CSeq: 16669 REGISTER From: "9167600147" <sip:9167600147@sip.broadvoice.com>;tag=2008174283 To: "9167600147" <sip:9167600147@sip.broadvoice.com> Via: SIP/2.0/UDP 38.102.41.250:5060;branch=z9hG4bK-cb476705edf7fe4d563c3ca1fdb0f244 Contact: <sip:9167600147@38.102.41.250:5060> Expires: 30 Content-Length: 0
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Hello,
Pbxnsip registers with a "200 OK" to broadvoice. So registration is fine. I am however, getting "481" on my SIP phone whenever I call ANY number, area code or not.
On BroadVoice's admin section, I'm using BYOD, and for the "device", we have it set to "Asterisk 8-port". This is what they recommended as there is no Pbxnsip option.
I followed this: http://support.pbxnsip.com/index.php?title...amp;redirect=no
Using the latest Pbxnsip version: 4.0.1.3499 (Linux)
Lets pretend my 10-digit broadvoice phone number is: 9161234567
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Name: BroadVoice
Type: SIP Registration
Direction: Inbound/Outbound
Trunk Destination: Generic SIP Server
Display Name: 9161234567
Account: 9161234567
Domain: sip.broadvoice.com
Username: 9161234567
Password: [pass given by broadvoice]
Proxy Address: proxy.nyc.broadvoice.com:5060
Strict RTP Routing: No
Avoid RFC4122 (UUID): Yes
Accept Redirect: No
------------------------------------------------------
My dial plan using BroadVoice as a trunk (text version):
100;BroadVoice;;916;
110;BroadVoice;;1916;
120;BroadVoice;;xxxxxxxxxx;1*
130;BroadVoice;;([0-9]*)@.*;"sip:\1@\r;user=phone"
Any advice on this? It's starting to look like I should just dump BroadVoice, but I'd like to try and get it to work.
Ryan
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Hello,
We purchased a Grandstream GXW-4108 gateway, and want to use it as simply a gateway to the PSTN.
We have 1 line plugged into channel 1, and want Pbxnsip to connect to the gxw4108 to make our outgoing calls on the PSTN. I've tried a ton of configuration methods, and cannot seem to get this thing working with pbxnsip, tried both adding it as a SIP Gateway and SIP Registration in Pbxnsip.
Anyone have experience with this product? Otherwise, can anyone recommend a gateway for the PSTN they are successfully using with Pbxnsip?
Thanks,
Ryan
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Hello,
We are looking at the TDM1600 16 Port FXO/FXS PCI Card, for plugging analog lines into. Would this card work with Pbxnsip? If not, are there any other suggestions for this type of card?
Thanks in advance,
Ryan
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I can confirm this...
Even after setting the correct login credentials in pbx.xml, and testing the permissions manually, the pbx still cannot write to the database, unless I start MySQL with --skip-grant-tables. This is NOT feasible as noone wants to run MySQL without any kind of user authentication.
When will this be fixed?
Ryan
Showing correct called ID
in Agent Group Setup
Posted
I'm not talking about the From header, that shows just fine; I'm talking about the To header, as in someone called in TO the auto attendant, then hit the Agent Group, then hit an Agent. I want the "to" to actually show the agent they connected to, not just keep showing that they're on the auto attendant.