koolandrew
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Posts posted by koolandrew
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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
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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.
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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
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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.
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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.
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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
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Are there any plans to answer a call as i see the notification, but nothing happens when you hit the green button.
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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.
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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
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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.
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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
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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?
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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.
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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
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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.
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Sip logging is enabled, what do you mean exactly?
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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
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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.
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Is this codec actually available as i cannot get it to work.
Thanks
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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
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I am not sure if you posted it incorrectly but the ip route should be 192.x instead of 92.x. And just making sure that is the actual local ip of the pbx and if so, how did set it up as both the public and private ip should be an alias for the localhost when you set it up. This setup wouldn't have been necessary when you had the pstn setup.
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We converted the private key as suggested, imported both our wildcard certificate and the private key as "Server certificate chain + private key", rebooted the PBX and it worked!
Thanks you so much for bearing with us, problem solved! Kudos!
I think the problem is that the key is a PKCS#8 key, while the PBX expects a PKCS#1 key. That is the difference between BEGIN RSA PRIVATE KEY and BEGIN PRIVATE KEY.
We will add code that in the next version the PBX will "eat" both versions. For now, please convert the #8 into #1 using openssl:
openssl rsa -in pkcs8.pem -out pkcs1.pem
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We have used this setup for our website using apache, on routers, our mail server, but we cannot get it to work on the pbx, and it doesnt appear many others can either.
Please just come clean with me, if you cant do it, you cant do it. I get the feeling that you know this already so just let me know and i will move on.
Dial plan--removing non numeric characters
in Call Treatment
Posted
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.................