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iqe


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Fragezeichen Is Engin not a compatable VoIP provider? Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

I believe I have all my setting set correctly but my engin extension doesn't appear to be working. On the status page, all is green and but then the engin extension and associated trunk gets greyed out with a little circle on it after a few seconds. The queue gets circle above it. The log is as follows:

cdr_addon_mysql.c: MySQL database table not specified. Assuming "cdr"
chan_sip.c: Failed to authenticate on REGISTER to ';tag=as3e8a4478' (tries '3')
chan_sip.c: -- Registration for '0733331918@byo.engin.com.au' timed out, trying again

I have had the account for a couple of months now sitting 'idle'. Not quite sure what to try next. Does anyone have any suggestions? Than you for any assistance.

31.01.2008 22:33 iqelectrical is offline Search for Posts by iqelectrical Add iqelectrical to your Buddy List
ozp


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Are you sure you are registering with the exact correct username and password from Engin?

"Failed to authenticate" suggests an incorrect username or password (Note that some VoIP providers use a different username for authentication purposes).

01.02.2008 00:22 ozpoole is offline Search for Posts by ozpoole Add ozpoole to your Buddy List
Dia
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A couple of questions, regarding your setup. Have you registered with a SIP User Agent (soft-phone, ATA, IP phone) to the extension in question?

Have you tried to register with a soft-phone or ATA using the credentials (username, password, SIP Proxy) of the trunk in question?

01.02.2008 01:02 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
iqe


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Hmm Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

OZpoole:
I believe so. As per the wizard. But something I thought of note from log.

Registration for '0733331918@byo.engin.com.au' timed out, trying again

but the Packetman says the Register String should be something like this:

0733331918:my password goes here @byo.engin.com.au/0733331918

Could this be an issue?

Diafora:

I currently have a SPA3000 ATA working using those credentials (but it is regularly diverted to my mobile and is costing me to much money doing it this way).

01.02.2008 07:03 iqelectrical is offline Search for Posts by iqelectrical Add iqelectrical to your Buddy List
i-p
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Is Engin not a compatable VoIP provider? Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Please try naming your trunk 0733331918.

02.02.2008 12:42 i-p-tel is offline Search for Posts by i-p-tel Add i-p-tel to your Buddy List
iqe


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Fragezeichen One step forward and one back Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

OK. Renaming the trunk now allows a softphone to register. When I call it though, it goes straight to my engin voicemail (diversion is set to none at engin). I can make outgoing calls though. BUT when I try to use the same credentials on the SPA3000, it wont register. I had the Packetman take a look but he can see what the issue may be. He was replicating my faults at his end too so I guess that should rule out my hardware and ISP!? I have fixed the password issue. I have also deleted all the details and redone just with just the SIP. Changed the extenstion to 200 instead of 3333.

Also, what does this mean in the log:

Feb 4 18:55:14 NOTICE[28715] cdr.c: CDR simple logging enabled.
Feb 4 18:55:14 VERBOSE[28715] logger.c: == MySQL RealTime reloaded.
Feb 4 18:55:14 NOTICE[28715] indications.c: Removed default indication country 'us'
Feb 4 18:55:14 NOTICE[28715] cdr_addon_mysql.c: MySQL database table not specified. Assuming "cdr"

I repeats about every minute or so.

Thanks again for any assistance.

This post has been edited 1 time(s), it was last edited by iqe on 05.02.2008 at 00:45.

05.02.2008 00:43 iqelectrical is offline Search for Posts by iqelectrical Add iqelectrical to your Buddy List
iqe


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SPA Settings? Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

I have been trying all sorts of different configurations and my money is on the SPA3000 being the issue. Packetman used a SPA as well to test. I can only assume it is a setting in there somewhere. I have trolled the Internet trying all sort of different suggestions and settings but to no avail. If I can't get this trial set-up working, I guess I will have to cancel my subscription and go with someone local in Aus to host all the company lines. It would be a shame to give up if it was just a button I had to click in the settings of the SPA or the like......

09.02.2008 00:19 iqelectrical is offline Search for Posts by iqelectrical Add iqelectrical to your Buddy List
Dia
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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?

10.02.2008 06:04 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
iqe


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Would you believe only one. All I want to do is have my mobile phone ring at the same time as my ATA so I can answer the incoming call on either (if I am in the office or out on the road). As the business grows and my trust in this VoIP config grows, I would like to have more complex set-ups more in keeping with a call centre style.

I have inserted the custom strings in for dial tone, busy tone, etc as discovered thru the engin forum BUT when the ATA is registered directly to engin, these tones make no difference to the functionality of the device to make calls etc. They just make it easier to know what is happening (eg. busy tone as opposed to disconnect tone).

I also have a couple of SPA 2k if they are any easier to config?

12.02.2008 22:18 iqelectrical is offline Search for Posts by iqelectrical Add iqelectrical to your Buddy List
Dia
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If what you described in the first paragraph, is what you are trying to achieve, then I believe that using the FXO port of an SPA3K is an overkill.

On the other hand, ringing your mobile phone (Classic extension) at the same time with a SIP URI (SIP extension), can lead to synchronization issues between the calls.

What I propose in your case, is to setup a ring-group with one or two SIP extensions in the "extension list" and a Classic extension as the "Destination if no answer". That will allow an incoming call to a DID, to ring an extension for 10-15 seconds and then ring-through to your mobile phone via a trunk.

The above setup has an additional benefit, that in the event of losing Internet connectivity in your office, the incoming calls will bypass the SIP extensions and ring on your mobile phone or any other Classic Extension.

13.02.2008 16:45 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
mob


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RE: Is Engin not a compatable VoIP provider? Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

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

18.02.2010 12:43 mobiletelecom is offline Search for Posts by mobiletelecom Add mobiletelecom to your Buddy List
Dia
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RE: Is Engin not a compatable VoIP provider? Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

If I remember well, we had discussed this at length some time ago. SIP Re-Invites are hard to achieve, since they require either Public IPs at both ends or perfect NAT traversal. ITSPs avoid them like the plague, since they know 99% of their customers have NATed Private IPs assigned to their SIP UAs, and they will complain to the ITSP's support personnel about one-way voice.

To make a long story short, unless you can find similar statements with the word RE-invite or re-invite for each of your calls in question (all these excerpts are from a single call), no matter what PennyTel claims, the RTP stream traveled the Pacific Ocean twice.

Feb 19 10:43:52 VERBOSE[27129] chan_sip.c: SIP response 100 to RE-invite on outgoing call 2f93b3703546443a4377674735baa05f@xxx.xxx.xxx.xxx (Public IP of ITSP's SIP Proxy)
Feb 19 10:43:52 VERBOSE[27129] chan_sip.c: SIP response 200 to RE-invite on outgoing call 2f93b3703546443a4377674735baa05f@xxx.xxx.xxx.xxx (Public IP of ITSP's SIP Proxy)
.....
Feb 19 10:43:53 VERBOSE[27129] chan_sip.c: SIP response 200 to RE-invite on outgoing call daf2c214-f94a1483@xxx.xxx.xxx.xxx (Public IP of SIP UA)
.....
Feb 19 10:44:10 VERBOSE[27129] chan_sip.c: Got a SIP re-invite for call 2f93b3703546443a4377674735baa05f@xxx.xxx.xxx.xxx (Public IP of ITSP's SIP Proxy)
.....
Feb 19 10:44:11 VERBOSE[27129] chan_sip.c: SIP response 200 to RE-invite on outgoing call daf2c214-f94a1483@xxx.xxx.xxx.xxx (Public IP of SIP UA)

Even if I have tried to enable SIP Re-Invites on all my trunks, it is almost impossible to get the ITSP techs to allow them. So tread lightly.

19.02.2010 11:10 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
mob


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RE: Is Engin not a compatable VoIP provider? Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

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?

02.03.2010 08:21 mobiletelecom is offline Search for Posts by mobiletelecom Add mobiletelecom to your Buddy List
Dia
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RE: Is Engin not a compatable VoIP provider? Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

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.

02.03.2010 11:56 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
mob


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RE: Is Engin not a compatable VoIP provider? Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

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

02.03.2010 20:44 mobiletelecom is offline Search for Posts by mobiletelecom Add mobiletelecom to your Buddy List
Dia
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RE: Is Engin not a compatable VoIP provider? Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Regarding colocation in Australia, the monthly bandwidth limit is too restrictive, and there is no indication how much the extra capacity will cost. At one of the current data-centers in Europe, the limit is 2000 GB /month and the extra charge is: 14,90€ / per TB.

Regarding Pennytel's answer, are the SIP Re-Invites activayed on your account with them? If they are, then try a few outbound calls, and verify whether you can see RE-invite statements on your System Log.

Whether you see them or not, paste the appropriate log excepts from each test call, and we will try to figure it out.

03.03.2010 02:15 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
mob


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RE: Is Engin not a compatable VoIP provider? Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

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

06.03.2010 01:13 mobiletelecom is offline Search for Posts by mobiletelecom Add mobiletelecom to your Buddy List
Dia
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RE: Is Engin not a compatable VoIP provider? Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Since the issue only exists on trunk to trunk calls, can you please verify that on inbound calls which are answered by a SIP UA, that Re-Invites actually appear in your System Log?

17.03.2010 20:56 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
 
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