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Thread: RE: Busy Lamp Field / GXP-2000
mob

Replies: 5
Views: 34413

RE: Busy Lamp Field / GXP-2000 19.09.2010 01:15 Forum: Terminal Equipment

I'm using a Yealink SIP-T28P IP handset and this handset also gives the option to answer the extension if it is flashing - but in order for this feature to work it asks for the "Pickup Number" - does anyone know what this is for PBXes?

Thread: RE: Phone book
mob

Replies: 5
Views: 17542

RE: Phone book 13.06.2010 21:51 Forum: Feature Requests

I'm using Yealink SIP-T28P IP Phones which support remote URL phone book.

What would the URL be to access our phonebook remotely?

Thread: RE: Is Engin not a compatable VoIP provider?
mob

Replies: 17
Views: 44794

RE: Is Engin not a compatable VoIP provider? 06.03.2010 01:13 Forum: Providers

I have spoken with both www.Gotalk.com.au & www.Pennytel.com and they both said that re-invite are automatically enable on every account.

I thought it may have something to do with NAT setting in my router (by the way I found a great site www.portforward.com) but then I realised that I'm do not have any hardware or softphones connected when I am having the issue with re-invites when my trunk is calling my mobile extension as a classic extension when my SIP extension is not answering after 15 seconds - as I don't have the extension registered when I may experiencing the issue and doing the test calls.

So if my TSP supports re-invites then it must be to do with how I have the trunk, incoming routing, outgoing routing, extension or ringing group configured with PBXes.

Below is my system log of the test call:-

Mar 6 09:55:12 VERBOSE[16632] logger.c: Found RTP audio format 96
Mar 6 09:55:12 VERBOSE[16632] logger.c: Found RTP audio format 97
Mar 6 09:55:12 VERBOSE[16632] logger.c: Found RTP audio format 18
Mar 6 09:55:12 VERBOSE[16632] logger.c: Found RTP audio format 4
Mar 6 09:55:12 VERBOSE[16632] logger.c: Found RTP audio format 8
Mar 6 09:55:12 VERBOSE[16632] logger.c: Found RTP audio format 0
Mar 6 09:55:12 VERBOSE[16632] logger.c: Found RTP audio format 101
Mar 6 09:55:12 VERBOSE[16632] logger.c: Peer audio RTP is at port 202.169.178.12:16778
Mar 6 09:55:12 VERBOSE[16632] logger.c: Peer video RTP is at port 202.169.178.12:65535
Mar 6 09:55:12 VERBOSE[16632] logger.c: Found description format iLBC
Mar 6 09:55:12 VERBOSE[16632] logger.c: Found description format iLBC
Mar 6 09:55:12 VERBOSE[16632] logger.c: Found description format G729
Mar 6 09:55:12 VERBOSE[16632] logger.c: Found description format G723
Mar 6 09:55:12 VERBOSE[16632] logger.c: Found description format PCMA
Mar 6 09:55:12 VERBOSE[16632] logger.c: Found description format PCMU
Mar 6 09:55:12 VERBOSE[16632] logger.c: Found description format telephone-event
Mar 6 09:55:12 VERBOSE[16632] logger.c: Capabilities: us - 0x71e (gsm|ulaw|alaw|g726|g729|speex|ilbc), peer - audio=0x50d (g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50c (ulaw|alaw|g729|ilbc)
Mar 6 09:55:12 VERBOSE[16632] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Mar 6 09:55:13 VERBOSE[30141] logger.c: -- Called 0400XXXXXX@from-internal/n
Mar 6 09:55:13 VERBOSE[30155] logger.c: We're at 124.108.37.109 port 43682
Mar 6 09:55:13 VERBOSE[30155] logger.c: Video is at 124.108.37.109 port 38402
Mar 6 09:55:13 VERBOSE[30155] logger.c: Adding codec 0x4 (ulaw) to SDP
Mar 6 09:55:13 VERBOSE[30155] logger.c: Adding codec 0x8 (alaw) to SDP
Mar 6 09:55:13 VERBOSE[30155] logger.c: Adding codec 0x10 (g726) to SDP
Mar 6 09:55:13 VERBOSE[30155] logger.c: Adding codec 0x400 (ilbc) to SDP
Mar 6 09:55:13 VERBOSE[30155] logger.c: Adding codec 0x2 (gsm) to SDP
Mar 6 09:55:13 VERBOSE[30155] logger.c: Adding codec 0x200 (speex) to SDP
Mar 6 09:55:13 VERBOSE[30155] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Mar 6 09:55:13 VERBOSE[30155] logger.c: -- Called Pennytel075313XXXX/0400XXXXXX
Mar 6 09:55:13 VERBOSE[16632] chan_sip.c: SIP response 100 to standard invite
Mar 6 09:55:13 VERBOSE[16632] chan_sip.c: SIP response 401 to standard invite
Mar 6 09:55:13 VERBOSE[16632] logger.c: We're at 124.108.37.109 port 43682
Mar 6 09:55:13 VERBOSE[16632] logger.c: Video is at 124.108.37.109 port 38402
Mar 6 09:55:13 VERBOSE[16632] logger.c: Adding codec 0x4 (ulaw) to SDP
Mar 6 09:55:13 VERBOSE[16632] logger.c: Adding codec 0x8 (alaw) to SDP
Mar 6 09:55:13 VERBOSE[16632] logger.c: Adding codec 0x10 (g726) to SDP
Mar 6 09:55:13 VERBOSE[16632] logger.c: Adding codec 0x400 (ilbc) to SDP
Mar 6 09:55:13 VERBOSE[16632] logger.c: Adding codec 0x2 (gsm) to SDP
Mar 6 09:55:13 VERBOSE[16632] logger.c: Adding codec 0x200 (speex) to SDP
Mar 6 09:55:13 VERBOSE[16632] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Mar 6 09:55:13 VERBOSE[16632] chan_sip.c: SIP response 100 to standard invite
Mar 6 09:55:18 VERBOSE[16632] chan_sip.c: SIP response 183 to standard invite
Mar 6 09:55:18 VERBOSE[16632] logger.c: Found RTP audio format 0
Mar 6 09:55:18 VERBOSE[16632] logger.c: Found RTP audio format 101
Mar 6 09:55:18 VERBOSE[16632] logger.c: Peer audio RTP is at port 202.85.241.98:20608
Mar 6 09:55:18 VERBOSE[16632] logger.c: Peer video RTP is at port 202.85.241.98:65535
Mar 6 09:55:18 VERBOSE[16632] logger.c: Found description format PCMU
Mar 6 09:55:18 VERBOSE[16632] logger.c: Found description format telephone-event
Mar 6 09:55:18 VERBOSE[16632] logger.c: Capabilities: us - 0x71e (gsm|ulaw|alaw|g726|g729|speex|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Mar 6 09:55:18 VERBOSE[16632] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Mar 6 09:55:18 VERBOSE[30141] logger.c: We're at 124.108.37.109 port 42178
Mar 6 09:55:18 VERBOSE[30141] logger.c: Video is at 124.108.37.109 port 37406
Mar 6 09:55:18 VERBOSE[30141] logger.c: Adding codec 0x4 (ulaw) to SDP
Mar 6 09:55:18 VERBOSE[30141] logger.c: Adding codec 0x8 (alaw) to SDP
Mar 6 09:55:18 VERBOSE[30141] logger.c: Adding codec 0x400 (ilbc) to SDP
Mar 6 09:55:18 VERBOSE[30141] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Mar 6 09:55:18 VERBOSE[30141] chan_sip.c: Oooh, format changed to 1024
Mar 6 09:55:18 VERBOSE[30141] chan_sip.c: Oooh, format changed to 4
Mar 6 09:55:27 VERBOSE[16632] chan_sip.c: SIP response 200 to standard invite
Mar 6 09:55:27 VERBOSE[16632] logger.c: Found RTP audio format 0
Mar 6 09:55:27 VERBOSE[16632] logger.c: Found RTP audio format 101
Mar 6 09:55:27 VERBOSE[16632] logger.c: Peer audio RTP is at port 202.85.241.98:20608
Mar 6 09:55:27 VERBOSE[16632] logger.c: Peer video RTP is at port 202.85.241.98:65535
Mar 6 09:55:27 VERBOSE[16632] logger.c: Found description format PCMU
Mar 6 09:55:27 VERBOSE[16632] logger.c: Found description format telephone-event
Mar 6 09:55:27 VERBOSE[16632] logger.c: Capabilities: us - 0x71e (gsm|ulaw|alaw|g726|g729|speex|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Mar 6 09:55:27 VERBOSE[16632] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Mar 6 09:55:27 VERBOSE[30155] logger.c: -- SIP/Pennytel075313XXXX-bc19 answered Local/0400XXXXXX@from-internal/n-f070,2
Mar 6 09:55:27 VERBOSE[30141] logger.c: -- Local/0400XXXXXX@from-internal/n-f070,1 answered SIP/09410719-a375
Mar 6 09:55:27 VERBOSE[30141] logger.c: We're at 124.108.37.109 port 42178
Mar 6 09:55:27 VERBOSE[30141] logger.c: Video is at 124.108.37.109 port 37406
Mar 6 09:55:27 VERBOSE[30141] logger.c: Adding codec 0x4 (ulaw) to SDP
Mar 6 09:55:27 VERBOSE[30141] logger.c: Adding codec 0x8 (alaw) to SDP
Mar 6 09:55:27 VERBOSE[30141] logger.c: Adding codec 0x400 (ilbc) to SDP
Mar 6 09:55:27 VERBOSE[30141] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Mar 6 09:55:38 VERBOSE[30155] chan_sip.c: Hangup call SIP/Pennytel075313XXXX-bc19, SIP callid 020cfc98706690b830df43e9321b41a1@sip.pennytel.com
Mar 6 09:55:38 VERBOSE[30141] chan_sip.c: Hangup call SIP/09410719-a375, SIP callid 349f-46f-252010235512-TCP_IMG02-1-210.80.190.195

Thread: RE: Is Engin not a compatable VoIP provider?
mob

Replies: 17
Views: 44794

RE: Is Engin not a compatable VoIP provider? 02.03.2010 20:44 Forum: Providers

I have spoken with Pennytel and they said that they support re-invites, and to configure setting and any issue to let them know support@pennytel.com - so why would my calls not be re-inviting - I am using softphones (X-lite & Eyebeam 1.5)

Also just for your info - Colocation server in Australia costs AU$119 per month for 1RU 20GB data

Thread: RE: Is Engin not a compatable VoIP provider?
mob

Replies: 17
Views: 44794

RE: Is Engin not a compatable VoIP provider? 02.03.2010 08:21 Forum: Providers

Based on this - would that mean that this solution would not be suitable for me as I'm based in Australia - will it also be suitable when PBXes has a server in or near Australia?

Is there any other way I could get the calls from myy extension to my mobile without call quality issues?

Thread: RE: Is Engin not a compatable VoIP provider?
mob

Replies: 17
Views: 44794

RE: Is Engin not a compatable VoIP provider? 18.02.2010 12:43 Forum: Providers

I have tried this solution Diafora, but the incoming caller experiences call quality issues when the call diverts to my mobile (Classic Extension) - I have set audio bypass to YES and reinvites are supported by my SIP provider. I am in Australia. My SIP provider is based in Australia. The classic extension is based in Australia. The incoming caller is based in Australia and the PBXes server is based in Tokyo.

Can you make any other suggestions to help improve the quality of the calls for the incoming caller (it sounds fine on my end).

The system log for a test call is below:-

Feb 18 21:20:57 VERBOSE[3266] logger.c: -- SIP/Pennytel0753133333-a492 answered Local/0402190000@from-internal/n-775a,2
Feb 18 21:20:57 VERBOSE[3253] logger.c: -- Local/0402190000@from-internal/n-775a,1 answered SIP/09410719-58d6
Feb 18 21:20:57 VERBOSE[3253] logger.c: We're at 124.108.37.109 port 41174
Feb 18 21:20:57 VERBOSE[3253] logger.c: Video is at 124.108.37.109 port 44278
Feb 18 21:20:57 VERBOSE[3253] logger.c: Adding codec 0x4 (ulaw) to SDP
Feb 18 21:20:57 VERBOSE[3253] logger.c: Adding codec 0x8 (alaw) to SDP
Feb 18 21:20:57 VERBOSE[3253] logger.c: Adding codec 0x400 (ilbc) to SDP
Feb 18 21:20:57 VERBOSE[3253] logger.c: Adding non-codec 0x1 (telephone-event) to SDP

Thread: Issue registering new Australian account
mob

Replies: 0
Views: 8335

Issue registering new Australian account 15.02.2010 12:55 Forum: Miscellaneous

I am trying to register a new account so I can then resell - but when I sign up I get the same error - "Invalid captcha" and no account has been created - do you know why this may be happening?

Thread: RE: Adding multiple extension to use outgoing route
mob

Replies: 2
Views: 15499

Adding multiple extension to use outgoing route 15.12.2009 20:00 Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

How do you add multiple extension to use an outgoing route. It meantions using X for wildcard, but that doesn't make any sense to me. Can you please explain with examples of what I need to type and in what field for this to work. Thanks

Thread: RE: Incoming calls hang-up when answered on Eyebeam 1.5 softphone
mob

Replies: 3
Views: 17291

RE: Incoming calls hang-up when answered on Eyebeam 1.5 softphone 02.12.2009 23:59 Forum: Terminal Equipment

I appreciate the support you have provided in the past, however I did log a case and it wasn't resolved completely as I hadn't replied with a response to your suggestion.

Support Case - 10/27/2009 12:25:52 AM Eyebeam 1.5 Softphone Call Hang-Up Issue

Thread: RE: Ring to Mobile Extension when Softphone (SIP) Extension is not Active
mob

Replies: 1
Views: 8322

Ring to Mobile Extension when Softphone (SIP) Extension is not Active 02.12.2009 20:09 Forum: Miscellaneous

I would like to be able to divert the calls to my mobile phone when the incoming caller rings my SIP DID for chooses my extension from digital receptionist - ONLY when my softphone SIP client is switched off on my laptop.

Thread: RE: Incoming calls hang-up when answered on Eyebeam 1.5 softphone
mob

Replies: 3
Views: 17291

Incoming calls hang-up when answered on Eyebeam 1.5 softphone 02.12.2009 20:01 Forum: Terminal Equipment

I am using Counterpath Eyebeam 1.5 softphone, it was working fine and I haven't changed the settings, but now when answering an incoming call the call is disconnected immediately, so I have to wait for the call to go to voicemail and then ring them back.

Can you please help with this issue?

P.S. I paid for personal support, and got a response, but I hadn't replied yet and the support was closed and I had no more credit - I don't mind paying for support if I actually get it!!!

Thread: Predictive Dialer
mob

Replies: 0
Views: 6459

Predictive Dialer 30.07.2009 04:12 Forum: Feature Requests

It would be great to have a predictive dialer which was built-in to PBXes or integrated with another hosted predictive dialer.

Please let me know if this is on the roadmap or if PBXes is already compatible with another hosted predictive dialer.

Thanks for the great IP PBX.

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