koolandrew

We are doing that now and all it is is a replication of the first button. What we would like to have a button for every caller in the queue? We need any available methods to notify manage

Is there any way to show multiple calls on an agent group. We have a added a BLF button to phone, but it obviously doesnt show us the status if there are two calls. Therefore we added another button, but it only reflects if that button is busy. Would there be a way of showing the status of each call separately, this has come up do to another problem.

People do it out of habit, or they copied and pasted, and it is certainly easier to read than all numbers. However, these all do need to removed, as you already remove spaces by default... You need to leave the + alone, as removing it will work when it is +15555551212 but doesnt work if is +442075551212, as i dont believe there is a standard to the amount of digits that follow a country code, and now we have 3 digit country codes, so that is best left to handled in the dial plan. Either way, it works when calling from their phone using the native phone dialler but wont work when using a sip dilaller if they are calling from their phone book. thx

I am very surprised that this cannot be resolved. The last response is no help. So i would send out an email, hey customers please remove () and - from your phone numbers in your phone book. Even though it has been working in your phone, it will no longer work when using your pbx........................come on.................

thanks for the reply. I agree about the agent, but i would rather take care of it on the pbx or domain level. I have no idea to implement your last line, could you please provide an example of such.I would like to leave the + alone though, as it is properly handled at the moment, just the other situations with ( ) and -. Thanks

Hi, i have found that customers who are using softphones on their cell phones have issues with dialling through the pbx. For example if the number is stored in their phone book, in any of the ways below, i cannot determine how to ignore or remove the non-numeric characters: 1(555)555-1212 or 555-555-1212 or (555) 555 1212 If there are any type of bracket that they might have used in their phone book, i cannot figure out how to create a rule in the dial plan that ignores those characters. Please help.

Yes, 1.You need to create an ivr.node, lets say its account 800. 2. If you want all calls to flow through this, you will need to add all DID's that would be called as an alias to the account.. Let say the DID was 666-666-6000 ie Account number(s) 800 6666666000 3. You would need to create a from base routing match list: !5555555555!112 !.*!700----------------All calls from 5555555555 would go to account 112, all others would go to account 700 Good luck

As stated above, If i have a missed call from +155511212 and hit return, it will not work. If i have a missed call from +91123456789 and hit return, it will not work. When i receive a call on a softphone on my mobile device, so i am not sure what you mean by country codes. If we declare 1 as our country code, we get other problems.

I cannot figure this out for return calls. If i have a missed call from +155511212 and hit return, it will not work. If i have a missed call from +91123456789 and hit return, it will not work. I am using +1 => 1 and +* => 011* and neither work. Please help.

Can you please clarify your statement above. Webrtc doesnt work on any Apple products as they dont support it. I dont really understand your statement above Thanks

Are there any plans to answer a call as i see the notification, but nothing happens when you hit the green button.

HI this topic is almost four years old, but i would like to respond and expand the request. What kind of reports can we create as some of our clients are becoming more sophisticated and hiring people with agent mgmt experience and interested in the types of reports that we can provide. I cannot find any documentation on this other than what you wrote above. Please advise.

I have checked out your latest version, and it works on chrome on desktop. Regarding logging in as a user from chrome on a mobile phone, or on android, i see that an image of a phone automatically appears, However, there is a media error, maybe because there is no plugin, and you probably know this already, although i dont know what "Chrome on mobile devices also does it" means from your comment above. Would you happen to know if you think you will continue to use the same setup as outlined in the usr_phone.js files? Thanks

That is an interesting update. I have gone to your website for an update, and there is some info. 1. What version of pbx is it supported from? As i cannot seem to find the release notes anymore, but i remember reading that there are some changes recently to how chrome and mozilla handle webrtc. For example, is it only from version 5.4 and on? 2. You claim webrtc is standard, for a desktop, not a mobile browser, as i dont believe there is any way to install extensions on a mobile browser. Is there anything for a mobile browser as of yet, or is that on the horizon? Therefore, my question remains. Is there an api that we could utilize to try and create our own app/webpage such that we could use for webrtc. Again, i dont want to keep bugging you, and if that is it, so be it.

Thanks for responding. To be honest, i dont think we have ever touched the codecs until now, as i was trying to explore better options for mobile devices. I saw your article on G726, and i thought i would try it. In my travels, i have learned about opus, a free codec, and wondered why you havent added it to the list. Thanks

I dont really understand your response. I never changed any trunk settings, so again, why cant we use g726 when the device is using it, but the pbx doesnt recognize it. Regarding OPUS being on our list, does that mean it is coming?

HI, It is almost 1.5 years since my last post, as we didnt get a response. We would still like to pursue this using vodia, but i dont know to do so. Other pbx platforms are embracing this technology along with video, so i am just wondering if vodia plans on the same. I realize that this may be outside of your core, and it is really only meant for talk buttons on websites, and that is fine. I am just wondering if there have been any updates on Vodia's side to support Webrtc calling on the extension level.

I am not clear on this as the domain trunks generally dont differ with the settings for the system. Can we add other codecs to the pbx like opus, is that possible. Thanks

8] 2016/04/19 15:36:44: Tagging request with existing tag [8] 2016/04/19 15:36:44: Port 56: Added rtpmap codec 19 (mapped to 116) [8] 2016/04/19 15:36:44: Port 56: Added rtpmap codec 5 (mapped to 115) [8] 2016/04/19 15:36:44: Port 56: Added rtpmap codec 21 (mapped to 114) [8] 2016/04/19 15:36:44: Port 56: Added rtpmap codec 20 (mapped to 113) [8] 2016/04/19 15:36:44: Port 56: Added rtpmap codec 1 (mapped to 101) [6] 2016/04/19 15:36:44: Port 56: Sending RTP to localpublicip.com:4000, codec not set yet [5] 2016/04/19 15:36:44: Port 56: Incoming call in domain vodiapbx.com on port 56 extension 121 [8] 2016/04/19 15:36:44: Call state for call object 467: idle [5] 2016/04/19 15:36:44: Port 56: New call created with number 467 [7] 2016/04/19 15:36:44: Port 56: Set codec preference count 1 [8] 2016/04/19 15:36:44: Call state for call object 467: connected [8] 2016/04/19 15:36:44: Port 56: state code from 0 to 200 [8] 2016/04/19 15:36:44: Port 56: Ignore double SDP [3] 2016/04/19 15:36:44: Port 56: Update codecs preference size 1, available codecs list is empty [5] 2016/04/19 15:36:44: Port 56: Available codec list is empty when trying to connect [8] 2016/04/19 15:36:44: Port 56: Send hangup with reason bye [5] 2016/04/19 15:36:44: Port 56: 30 seconds callback set for force cleanup [7] 2016/04/19 15:36:44: Messages in the call port 56 2016/4/19 15:36:44 Rx: udp:localpublicip.com:1072 (1249 bytes) INVITE sip:*97@vodiapbx.com SIP/2.0 Via: SIP/2.0/UDP localpublicip.com:1072;rport;branch=z9hG4bxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx Max-Forwards: 70 From: <sip:121@vodiapbx.com>;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx To: <sip:*97@vodiapbx.com> Contact: <sip:121@localpublicip.com:1072;ob> Call-ID: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx CSeq: 4281 INVITE Route: <sip:vodiapbx.com;transport=udp;lr> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_zerofltebmc-22/r2457 Content-Type: application/sdp Content-Length: 539 v=0 o=- 3670083402 3670083402 IN IP4 localpublicip.com s=pjmedia c=IN IP4 localpublicip.com t=0 0 m=audio 4000 RTP/AVP 116 115 114 113 101 c=IN IP4 localpublicip.com a=rtcp:4001 IN IP4 localpublicip.com a=sendrecv a=rtpmap:116 G726-40/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:114 G726-24/8000 a=rtpmap:113 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:mCpWmYI+907R4p0eX3xuy6sO/rVLLeSSjBfLSwcy a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:mnu3EI83ZCW0/Cipso5apz+JCmAkm46qL0EzE9F3 2016/4/19 15:36:44 Tx: udp:localpublicip.com:1072 (329 bytes) SIP/2.0 100 Trying Via: SIP/2.0/UDP localpublicip.com:1072;rport=1072;branch=z9hG4bxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx From: <sip:121@vodiapbx.com>;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx To: <sip:*97@vodiapbx.com>;tag=cadxxxxxx Call-ID: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx CSeq: 4281 INVITE Content-Length: 0 2016/4/19 15:36:44 Tx: udp:localpublicip.com:1072 (525 bytes) SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP localpublicip.com:1072;rport=1072;branch=z9hG4bxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx From: <sip:121@vodiapbx.com>;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx To: <sip:*97@vodiapbx.com>;tag=cadxxxxxx Call-ID: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx CSeq: 4281 INVITE User-Agent: Vodia-PBX/5.3.0 WWW-Authenticate: Digest realm="vodiapbx.com",nonce="173f278f4f6fe8da87cc5d8b89c66cf7",domain="sip:*97@vodiapbx.com",algorithm=MD5 Content-Length: 0 2016/4/19 15:36:44 Rx: udp:localpublicip.com:1072 (416 bytes) ACK sip:*97@vodiapbx.com SIP/2.0 Via: SIP/2.0/UDP localpublicip.com:1072;rport;branch=z9hG4bxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx Max-Forwards: 70 From: <sip:121@vodiapbx.com>;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx To: <sip:*97@vodiapbx.com>;tag=cadxxxxxx Call-ID: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx CSeq: 4281 ACK Route: <sip:vodiapbx.com;transport=udp;lr> Content-Length: 0 2016/4/19 15:36:44 Rx: udp:localpublicip.com:1072 (1457 bytes) INVITE sip:*97@vodiapbx.com SIP/2.0 Via: SIP/2.0/UDP localpublicip.com:1072;rport;branch=z9hG4bKPjRLpDgdP9I25MRXFt-1YKR1jdsRrQDDXc Max-Forwards: 70 From: <sip:121@vodiapbx.com>;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx To: <sip:*97@vodiapbx.com> Contact: <sip:121@localpublicip.com:1072;ob> Call-ID: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx CSeq: 4282 INVITE Route: <sip:vodiapbx.com;transport=udp;lr> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_zerofltebmc-22/r2457 Authorization: Digest username="121", realm="vodiapbx.com", nonce="173f278f4f6fe8da87cc5d8b89c66cf7", uri="sip:*97@vodiapbx.com", response="bdeacdc5725b2c3ae309c242b679d3e8", algorithm=MD5 Content-Type: application/sdp Content-Length: 539 v=0 o=- 3670083402 3670083402 IN IP4 localpublicip.com s=pjmedia c=IN IP4 localpublicip.com t=0 0 m=audio 4000 RTP/AVP 116 115 114 113 101 c=IN IP4 localpublicip.com a=rtcp:4001 IN IP4 localpublicip.com a=sendrecv a=rtpmap:116 G726-40/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:114 G726-24/8000 a=rtpmap:113 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:mCpWmYI+907R4p0eX3xuy6sO/rVLLeSSjBfLSwcy a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:mnu3EI83ZCW0/Cipso5apz+JCmAkm46qL0EzE9F3 2016/4/19 15:36:44 Tx: udp:localpublicip.com:1072 (329 bytes) SIP/2.0 100 Trying Via: SIP/2.0/UDP localpublicip.com:1072;rport=1072;branch=z9hG4bxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx From: <sip:121@vodiapbx.com>;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx To: <sip:*97@vodiapbx.com>;tag=cadxxxxxx Call-ID: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx CSeq: 4282 INVITE Content-Length: 0 2016/4/19 15:36:44 Tx: udp:localpublicip.com:1072 (562 bytes) SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP localpublicip.com:1072;rport=1072;branch=z9hG4bxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx From: <sip:121@vodiapbx.com>;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx To: <sip:*97@vodiapbx.com>;tag=cadxxxxxx Call-ID: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx CSeq: 4282 INVITE Contact: <sip:121@vodiapbxip:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Vodia-PBX/5.3.0 Content-Length: 0 2016/4/19 15:36:44 Rx: udp:localpublicip.com:1072 (416 bytes) ACK sip:*97@vodiapbx.com SIP/2.0 Via: SIP/2.0/UDP localpublicip.com:1072;rport;branch=z9hG4bxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx Max-Forwards: 70 From: <sip:121@vodiapbx.com>;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx To: <sip:*97@vodiapbx.com>;tag=cadxxxxxx Call-ID: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx CSeq: 4282 ACK Route: <sip:vodiapbx.com;transport=udp;lr> Content-Length: 0 [8] 2016/04/19 15:36:44: Port 56: Clearing port with SIP Call-ID xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx [6] 2016/04/19 15:36:44: Reg 9885, Sent MWI notification 5/0 (0/0) to user 121@vodiapbx.com [6] 2016/04/19 15:36:44: Reg 9949, Sent MWI notification 5/0 (0/0) to user 121@vodiapbx.com [9] 2016/04/19 15:36:44: Using outbound proxy sip:localpublicip.com:55463;transport=udp because of flow-label [9] 2016/04/19 15:36:44: SOAP: Store CDR in http://pbxcdr.com/cdr.php <env:Envelope xmlns:env="http://schemas.xmlsoap.org/soap/envelope/" xmlns:sns="http://soap.com/pbx"><env:Body><sns:CDR><PrimaryCallID>xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx</PrimaryCallID><CallID>xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx</CallID><From>"Kool Tel.com" <sip:121@vodiapbx.com></From><To><sip:*97@vodiapbx.com></To><Direction>I</Direction><Type>mailbox</Type><AccountNumber>121@vodiapbx.com</AccountNumber><RemoteParty>"Kool Tel.com" <sip:121@vodiapbx.com></RemoteParty><LocalParty>121</LocalParty><TrunkName></TrunkName><TrunkID></TrunkID><Domain>vodiapbx.com</Domain><LocalTime>20160419153644</LocalTime><TimeStart>20160419193644</TimeStart><Extension>121@vodiapbx.com</Extension><TimeConnected>20160419193644</TimeConnected><DurationHHMMSS>0:00:00</DurationHHMMSS><Duration>0</Duration><TimeEnd>20160419193644</TimeEnd><IPAdr>udp:localpublicip.com:1072</IPAdr><IdleDuration>1825</IdleDuration><Quality>VQSessionReport: CallTerm LocalMetrics: CallID:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx FromID:<sip:121@vodiapbx.com>;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx ToID:<sip:*97@vodiapbx.com>;tag=cadxxxxxx LocalAddr:IP=0.0.0.0 PORT=51640 SSRC=0x9xxxxxx RemoteAddr:IP=0.0.0.0 PORT=0 SSRC=0x x-UserAgent:Vodia-PBX/5.3.0 x-SIPterm:SDC=OK SDR=OR </Quality></sns:CDR></env:Body></env:Envelope> [8] 2016/04/19 15:36:44: Remove leg 1081: Call port 56, SIP call id xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx [7] 2016/04/19 15:36:44: http:pbxcdr.com:80: DNS A returned cdrserver.com [7] 2016/04/19 15:36:44: http:pbxcdr.com:80: Connect to cdrserver.com [9] 2016/04/19 15:36:47: http:pbxcdr.com:80: Send request POST /cdr.php HTTP/1.1 Host: pbxcdr.com Content-Length: 1405 SOAPAction: Trunk-CDR Content-Type: text/xml Accept-Language: en-us User-Agent: Mozilla/4.0 (compatible; PBX) <env:Envelope xmlns:env="http://schemas.xmlsoap.org/soap/envelope/" xmlns:sns="http://soap.com/pbx"><env:Body><sns:CDR><PrimaryCallID>xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx</PrimaryCallID><CallID>xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx</CallID><From>"Kool Tel.com" <sip:121@vodiapbx.com></From><To><sip:*97@vodiapbx.com></To><Direction>I</Direction><Type>mailbox</Type><AccountNumber>121@vodiapbx.com</AccountNumber><RemoteParty>"Kool Tel.com" <sip:121@vodiapbx.com></RemoteParty><LocalParty>121</LocalParty><TrunkName></TrunkName><TrunkID></TrunkID><Domain>vodiapbx.com</Domain><LocalTime>20160419153644</LocalTime><TimeStart>20160419193644</TimeStart><Extension>121@vodiapbx.com</Extension><TimeConnected>20160419193644</TimeConnected><DurationHHMMSS>0:00:00</DurationHHMMSS><Duration>0</Duration><TimeEnd>20160419193644</TimeEnd><IPAdr>udp:localpublicip.com:1072</IPAdr><IdleDuration>1825</IdleDuration><Quality>VQSessionReport: CallTerm LocalMetrics: CallID:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx FromID:<sip:121@vodiapbx.com>;tag=xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx ToID:<sip:*97@vodiapbx.com>;tag=cadxxxxxx LocalAddr:IP=0.0.0.0 PORT=51640 SSRC=0x9xxxxxx RemoteAddr:IP=0.0.0.0 PORT=0 SSRC=0x x-UserAgent:Vodia-PBX/5.3.0 x-SIPterm:SDC=OK SDR=OR </Quality></sns:CDR></env:Body></env:Envelope> [9] 2016/04/19 15:36:47: Received 319 bytes

Here you go, they have been sanitized per my customer's request.

Sip logging is enabled, what do you mean exactly?

This is what happens when i try to connect [3] 2016/04/19 14:55:35: Port 45: Update codecs preference size 1, available codecs list is empty [5] 2016/04/19 14:55:35: Port 45: Available codec list is empty when trying to connect I get an error 415 from the phone. I am not sure what else you need, please try it yourself. Thanks

I am not totally clear in what you have mentioned here but in my previous attempts to start this topic, i mentioned that i disabled all codecs but g726 in the pbx, then enabled all the g726 codecs on the csipsimple softphone, and i got a message on the pbx logs that there were no codecs available, so it appeared that the codec was not installed or enabled.

Is this codec actually available as i cannot get it to work. Thanks

I was interested in a low bandwidth codec, and i came across an old article about g726. I noticed that it is listed in the list of available codecs, but i tried to use it with several handsets and softphones, and when i look on the logs i get "unsupported media type" or not in the list of available codecs. I have tried many variants of g726, but even when i make it the only choice, i get the following: [3] 2016/04/14 12:54:25: Port 61: Update codecs preference size 1, available codecs list is empty [5] 2016/04/14 12:54:25: Port 61: Available codec list is empty when trying to connect It seems that the codec doesnt exist. Can this be resolved. thanks