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Thread: RE: Webcall
dor

Replies: 13
Views: 118231

RE: Webcall 22.07.2010 17:29 Forum: News

When I do it using my own form, it redirects the user to pbxes page. The last thing I want is to loose the user just like that. Is there any way so after submit it would sent the user back to the originating page?

Thread: RE: Call an URL on Phone Event
dor

Replies: 7
Views: 16121

RE: Call an URL on Phone Event 26.04.2010 23:01 Forum: Feature Requests

Another thing is HTTP POST as IVR option - it would enable triggering events using DTMF


I've already implemented similar functionality, but the other way around - any event on Micasaverde Vera z-wave controller can trigger PBXes webcall.

Thread: RE: Blacklist
dor

Replies: 3
Views: 15480

RE: Blacklist 06.03.2010 04:03 Forum: Feature Requests

Diafora, your idea is even better - to reuse the filter mechanism and syntax used for outbound routes and trunks - both regexp and plain listing. That would be perfect!

Thread: RE: Blacklist
dor

Replies: 3
Views: 15480

Blacklist 05.03.2010 17:58 Forum: Feature Requests

The one feature I'm really missing is blacklist - ability to have a list of CID for treating them in special way - sending them to VM or just hang up.
Using Incoming routes for this is a pain - if I add so many entries, the routes list will become too large to manage.

Any chance a blacklist can be implemented?

Thread: RE: Notification of Missed Calls
dor

Replies: 37
Views: 93035

RE: Notification of Missed Calls 05.03.2010 17:51 Forum: Feature Requests

Agree, it would very valuable

Thread: RE: PBXES down !!
dor

Replies: 3
Views: 7574

RE: PBXES down !! 01.12.2009 01:46 Forum: Bugs

Well, I can't access www4, phones don't work.

What's going on lately? I'm used to recommend pbxes as one of the most reliable option - now I'm starting to feel bad about it.

Thread: RE: www4 is down
dor

Replies: 5
Views: 10833

RE: www4 is down 22.11.2009 22:39 Forum: Bugs

That's understandable, it would be better though if you advise users on such situations just when detected, not after it's over. Twitter, email - anything is good as long as it let us take a workaround if necessary.

Thread: RE: www4 is down
dor

Replies: 5
Views: 10833

www4 is down 22.11.2009 03:55 Forum: Bugs

Getting this error on www4

[nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2)] ** mysql://asteriskuser:amp109@localhost/asterisk

Thread: RE: Reselling- Reseller's domain
dor

Replies: 15
Views: 98865

RE: Reselling- Reseller's domain 18.11.2009 07:57 Forum: News

Zitat:
Originally posted by i-p-tel
You should register to paid accounts only when using a custom domain as SIP server. There is no additional fee.

A SRV entry is not required for SIP servers. It's used for receiving calls by SIP URIs. You can add a SRV entry in the same fashion as the one of pbxes.org. This is really non trivial. By reading out ours you learn how to verify your own.


I'm trying to setup custom SIP URI.
Could you advise what SRV records you use: both both TCP (_sip._tcp) and UDP (_sip._udp_) ? Anything else, backup, etc?

Thread: RE: Phone can't connect to PBXes - www down as well?
dor

Replies: 40
Views: 79817

RE: Phone can't connect to PBXes - www down as well? 23.10.2009 06:54 Forum: Bugs

I support twitter notifications idea.
It's an excellent channel for this kind of things

Thread: RE: DNS issue again???
dor

Replies: 1
Views: 5353

DNS issue again??? 23.10.2009 06:47 Forum: Bugs

pbxes.ORG doesn't reply, while pbxes.COM works.

All my devices connect to .ORG - unsuccesfully

Thread: RE: Problem with incoming calls
dor

Replies: 7
Views: 22605

RE: Problem with incoming calls 09.10.2009 02:53 Forum: Providers

Zitat:
Originally posted by Diafora
• Do you still have it setup as a URI or a registered trunk?
• Can you ask VoIP.ms to send you another trace, now that is working?



Here's the new trace. Since then I switched back to registered trunk (I had SIP URI just as an unsuccessful attempt to cure this problem)

INVITE sip:5143330202@64.118.93.76:27436 SIP/2.0
Via: SIP/2.0/UDP 67.205.74.164:5060;branch=z9hG4bK00f21f6d;rport
From: "Steve P " <sip:8777864767@67.205.74.164>;tag=as0721813d
To: <sip:5143330202@64.118.93.76:27436>
Contact: <sip:8777864767@67.205.74.164>
Call-ID: 1cc4258369718d640a4612a1661d287c@67.205.74.164
CSeq: 102 INVITE
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "Steve P " <sip:8777864767@67.205.74.164>;privacy=off;screen=no
Date: Thu, 08 Oct 2009 18:28:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 30851 30851 IN IP4 67.205.74.164
s=session
c=IN IP4 67.205.74.164
t=0 0
m=audio 13702 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called 101108/5143330202
ca1*CLI>
<--- SIP read from 64.118.93.76:27436 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.205.74.164:5060;branch=z9hG4bK00f21f6d;received=67.205.74.164;rport=5060
From: "Steve P " <sip:8777864767@67.205.74.164>;tag=as0721813d
To: <sip:5143330202@64.118.93.76:27436>
Call-ID: 1cc4258369718d640a4612a1661d287c@67.205.74.164
CSeq: 102 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:5143330202@64.118.93.76:27436>
Content-Length: 0


<------------->

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.205.74.164:5060;branch=z9hG4bK00f21f6d;received=67.205.74.164;rport=5060
From: "Steve P " <sip:8777864767@67.205.74.164>;tag=as0721813d
To: <sip:5143330202@64.118.93.76:27436>;tag=as6a00ffa6
Call-ID: 1cc4258369718d640a4612a1661d287c@67.205.74.164
CSeq: 102 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:5143330202@64.118.93.76:27436>
Content-Length: 0


<------------->

Thread: RE: Phone can't connect to PBXes - www down as well?
dor

Replies: 40
Views: 79817

RE: Phone can't connect to PBXes - www down as well? 09.10.2009 02:01 Forum: Bugs

What's IP of www4?

Thread: RE: Problem with incoming calls
dor

Replies: 7
Views: 22605

RE: Problem with incoming calls 02.10.2009 04:51 Forum: Providers

Inbound route is there, and it shouldn't be the cause as originally the setup used registered trunk, not SIP URI. It worked OK, few days ago it broke, and I switched to SIP URI in attempt to troubleshoot the problem.

UPDATE: somehow it's back in track since yesterday. voip.ms said they didn't do anything, so I guess it'll stay a mistery...

Thread: RE: Problem with incoming calls
dor

Replies: 7
Views: 22605

Problem with incoming calls 29.09.2009 02:38 Forum: Providers

My main business line stopped working with no obvious cause.

Caller just hears short dial tones. On my side phones do ring after some delay, but by that time the caller is already disconnected.
I tried to monitor what's going on using Status screen: while caller is calling (extension is in red color) nothing rings on my side. After 10 seconds call disconnects by itself (extension icon goes green, caller gets short tones), then the phones start ringing. If I answer I obviously hear nothing as the caller is already disconnected.

In the log is see:
Sep 27 16:49:38 VERBOSE[1306] logger.c: -- Got SIP response 481 "Call Leg/Transaction Does Not Exist"

I opened a call in voip.ms support - they used to be highly reliable provider.


Here's what they say:
Zitat:

A SIp trace indicates that the problem seem to be located at PBXES, or its destination.

When we dial the number, it immediatly reaches our server, and tries to dial the sip uri you have set at pbxes. Pbxes sends back a "Trying" and nothing more, like if it was stuck, call is cancelled about 10 seconds later.

INVITE sip:doronin-0202@pbxes.org SIP/2.0
Via: SIP/2.0/UDP 67.205.74.164:5060;branch=z9hG4bK38b1433e;rport
From: "5146678178" ;tag=as4f7295fa
To:
Contact:
Call-ID: 4a6d81b56f17a45b008caa06267a0316@67.205.74.164
CSeq: 102 INVITE
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "5146678178" ;privacy=off;screen=no
Date: Mon, 28 Sep 2009 18:35:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 411

v=0
o=root 30851 30851 IN IP4 67.205.74.164
s=session
c=IN IP4 67.205.74.164
t=0 0
m=audio 14870 RTP/AVP 0 8 18 3 111 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called doronin-0202@pbxes.org
ca1*CLI>

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.205.74.164:5060;branch=z9hG4bK38b1433e
From: "5146678178" ;tag=as4f7295fa
To:
Call-ID: 4a6d81b56f17a45b008caa06267a0316@67.205.74.164
CSeq: 102 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact:
Content-Length: 0


P.S. I tried to send incoming call to both registered trunk and SIP URI with the same result.

Thread: Plugin for home automation controller
dor

Replies: 0
Views: 6263

Plugin for home automation controller 26.07.2009 02:14 Forum: Miscellaneous

Just in case you find it interesting...

I created a simple plugin for one of the most powerful home automation controllers on the market today - "Vera" from micasaverde.com; plugin allows to trigger webcall from z-wave handheld remotes and in-wall controllers, Vera's web UI, or *on events* from various timers, sensors, or other devices.

They're planning to make a repository for custom plugins in the nearest future, so I hope to publish it there.
VoIP Phone Dialer for Vera

Thread: RE: Android Phones over SIP
dor

Replies: 13
Views: 125026

RE: Android Phones over SIP 26.07.2009 00:15 Forum: News

What SIP port sipdroid listens to?
I'm curious because if it's 5060 and there are other SIP agents on LAN using the same port there may be conflicts.
(I know NAT *should* handle it, but there too many cases out there proving that home routers are not always good at it)

Thread: RE: Android Phones over SIP
dor

Replies: 13
Views: 125026

RE: Android Phones over SIP 23.07.2009 22:24 Forum: News

No, I just mean that currently sound from sipdroid can be sent to speaker, but not to BT headset.

Thread: RE: Android Phones over SIP
dor

Replies: 13
Views: 125026

RE: Android Phones over SIP 22.07.2009 05:56 Forum: News

Tried it on Rogers Canada 3G network, it works, but sound is very choppy, so here it seems to be usable on wifi only.

The real problem is lack of support for Bluetooth headsets. Any plans to add it?

Thread: Re: '+
dor

Replies: 3
Views: 8376

Re: '+ 15.07.2009 14:35 Forum: Miscellaneous

Thanks Diafora, it was late, so I added +| into wrong line... smile

Showing posts 1 to 20 of 125 results Pages (7): [1] 2 3 next » ... last »

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