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iqe
Registration Date: 01.01.1970
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31.01.2008 23:33 |
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i-p
Super Moderator
Registration Date: 14.01.2006
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02.02.2008 13:42 |
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iqe
Registration Date: 01.01.1970
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05.02.2008 01:43 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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Most probably a couple of settings need tweaking to get it to work properly. BUT the SPA-3K is the most complex PSTN gateway, to configure and get it to properly work.
The biggest obstacle in this process, is provisioning correctly the various regional settings, which differ from country to country, and even from telco to telco in some cases. Among these, are the normal and busy dial-tones, the disconnect tone, the CallerID method used etc.
So, using the regional settings of a GSM FWT in Italy, will most probably not help you disconnect properly from a PSTN phone line in Australia. The right settings are likely to be found from Australian users which have hooked these gateways to Telstra and Optus land-lines successfully.
How many phone lines do you plan to connect to SPA-3Ks, if you get them to work properly?
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10.02.2008 07:04 |
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mob
Registration Date: 01.01.1970
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18.02.2010 13:43 |
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Dia
Premium Account
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19.02.2010 12:10 |
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mob
Registration Date: 01.01.1970
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02.03.2010 09:21 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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RE: Is Engin not a compatable VoIP provider? |
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It would certainly be advantageous in your case, if PBXes could collocate a server in Australia, since the voice packets wouldn't have to travel across the ocean twice.
But, a collocated server would not resolve the issue, it would only alleviate its' extreme consequences. Asking Pennytel to allow SIP Re-Invites, would resolve the issue completely. SIP Re-Invites were conceived exactly for this purpose.
There is no technical reason, why the RTP stream has to be hair-pinned (proxied) via the ITSP's SIP Proxy, it is primarily done for NAT Traversal purposes. There are better ways to combat the one-way voice issues NAT causes. (STUN, ICE, TURN)
So contact Pennytel and talk it over with them, if they play hard ball (company policy, security implications) I am willing to talk to them as well. If after this, they still refuse to budge, you should start looking for another ITSP.
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02.03.2010 12:56 |
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mob
Registration Date: 01.01.1970
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02.03.2010 21:44 |
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Dia
Premium Account
Registration Date: 03.03.2006
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03.03.2010 03:15 |
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mob
Registration Date: 01.01.1970
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RE: Is Engin not a compatable VoIP provider? |
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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
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06.03.2010 02:13 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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17.03.2010 21:56 |
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