date:20060919

I wonder whether you are experiencing the following bug (since the SIP
INVITE has a multipart SDP body):
http://bugs.digium.com/view.php?id=7124nbn=4


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Dinesh Nair
 Sent: 19 September 2006 06:54
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] 488 Not acceptable here sent by 
 Asterisk - SIPdebug follows
 
 
 the situation
 
 Asterisk -- SIP --- SIPGW --- SIP Phone
 
 SIP Phone is trying to call asterisk dialplan:
 
 exten = 0224577501,1,Answer()
 exten = 0224577501,2,Playback(demo-instruct)
 exten = 0224577501,3,Hangup()
 
 however, asterisk 1.2.12.1 (on FreeBSD 6.1) sends back a 488 
 Not acceptable here with a CLI message of
 
 WARNING[162]: chan_sip.c:3529 process_sdp: Insufficient 
 information for SDP (m = '', c = '')
 
 
 it seems to be dropping out in process_sdp() because it can't 
 find the m= 
 or the c=. this is a little odd, so am wondering if this has 
 triggered some 
 edge case in find_sdp(), get_sdp() or get_sdp_iterate(). i've 
 been poring 
 thru the code (as the box is remote, and i cant duplicate it 
 locally), but 
 can't find exactly where in chan_sip.c its borking.
 
 any advice would be much appreciated.
 
 the SIP debug is attached below:
 
 (10.14.32.179 is the SIPGW, 10.14.32.164 is Asterisk)
 
   begin sip debug
 -- SIP read from 10.14.32.179:5060:
 INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
 Via: SIP/2.0/UDP 10.14.32.179:5060
 Via: SIP/2.0/UDP 10.14.32.189:5060
 Record-Route: sip:10.14.32.179:5060
 Supported: replaces
 User-Agent: SIP201 (lp201_sip0423.bin)
 Contact: sip:[EMAIL PROTECTED]:5060
 From: sip:[EMAIL PROTECTED]:5060 
 ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
 To: sip:[EMAIL PROTECTED]:5060;user=phone
 Call-ID: [EMAIL PROTECTED]
 CSeq: 1 INVITE
 History-Info: sip:[EMAIL PROTECTED]:5060;index 1
 Content-Type: multipart/mixed;boundary=unique-boundary
 Content-Length: 474
 
 --unique-boundary
 Content-Type: application/sdp
 
 v=0
 o=SIP201 12367 0 IN IP4 10.14.32.189
 s=SIP201 Session
 i=Audio Session
 c=IN IP4 10.14.32.189
 t=0 0
 m=audio 16384 RTP/AVP 4 18 0 8 18
 a=rtpmap:4 G723/8000/1
 a=rtpmap:18 G729/8000/1
 a=rtpmap:0 PCMU/8000/1
 a=rtpmap:8 PCMA/8000/1
 a=rtpmap:18 G729/8000/1
 
 --unique-boundary
 Content-Type: application/isup;version=Indonesia
 Content-Transfer-Encoding: binary
 
 
 --- (14 headers 21 lines)---
 Using INVITE request as basis request - 
 [EMAIL PROTECTED]
 Sending to 10.14.32.179 : 5060 (non-NAT)
 Found peer 'RISTI'
 Sep 19 09:38:53 WARNING[162]: chan_sip.c:3529 process_sdp: 
 Insufficient 
 information for SDP (m = '',
   c = '')
 Transmitting (no NAT) to 10.14.32.179:5060:
 SIP/2.0 488 Not acceptable here
 Via: SIP/2.0/UDP 10.14.32.179:5060;received=10.14.32.179
 Via: SIP/2.0/UDP 10.14.32.189:5060
 From: sip:[EMAIL PROTECTED]:5060 
 ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
 To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as5a7aa73d
 Call-ID: [EMAIL PROTECTED]
 CSeq: 1 INVITE
 User-Agent: QubeTalk ECS
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
 Contact: sip:[EMAIL PROTECTED]
 Content-Length: 0
 
 
 ---
 Destroying call '[EMAIL PROTECTED]'
 suria*CLI
 -- SIP read from 10.14.32.179:5060:
 ACK sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
 Via: SIP/2.0/UDP 10.14.32.179:5060
 Via: SIP/2.0/UDP 10.14.32.189:5060
 Record-Route: sip:10.14.32.179:5060
 Contact: sip:[EMAIL PROTECTED]:5060
 User-Agent: SIP201 (lp201_sip0423.bin)
 From: sip:[EMAIL PROTECTED]:5060 
 ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
 To: sip:[EMAIL PROTECTED]:5060;user=phone ;tag=as5a7aa73d
 Call-ID: [EMAIL PROTECTED]
 CSeq: 1 ACK
 Content-Length:0
 
 
 --- (11 headers 0 lines)---
 Destroying call '[EMAIL PROTECTED]'
   end sip debug
 
 
 -- 
 Regards,   /\_/\   All dogs go to heaven.
 [EMAIL PROTECTED](0 0)   
 http://www.openmalaysiablog.com/
 +==oOO--(_)--OOo==
 +
 | for a in past present future; do
 |
 |   for b in clients employers associates relatives 
 neighbours pets; do   |
 |   echo The opinions here in no way reflect the opinions of 
 my $a $b.  |
 | done; done  
 |
 +=
 +
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[asterisk-users] When does Scalability requests Asterisk to Use SER ?

2006-09-19 Thread Marco Mouta

Hi all,I'm planing to develop a solution based on Asterisk for about 300 users. My question now is, do I really need to use openSER as the sip proxy and Asterisk for the PBX functions?Can i trust in a solution only with Asterisk to make all this install?
Please help me with your experience on this kind of asterisk solutions.I've googled and read about asterisk at large scale solutions, but still in doubt.
http://www.voip-info.org/wiki-Asterisk+at+large-- Com os melhores cumprimentos,Marco Mouta
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Re: [asterisk-users] Cisco 7940 Problem (Mess)

2006-09-19 Thread Brian Candler

On Tue, Sep 19, 2006 at 12:12:03AM +0900, Gary Guthary wrote:
 This is going to be an exercise in 'Networking' for sure...
 
 The only catch is that per the phone's network settings:
 
 The phone uses a static IP of something like 192.168.0.220 with a Gateway of
 192.168.0.1. - Standard class 'C' netmask (255.255.255.0).
 
 The phone has DHCP DISABLED.
 
 The phone has it's TFTP server set to something like 62.120.xxx.xxx
 (something completely outside of the local network).
 
 My home LAN uses 10.0.0.xxx on the local side.
 
 But I can reset my XP-Box to 192.168.0.99 and ping the phone with no probs.
 
 But If I set my XP-Box to a static IP of the Phone's TFTP server, my
 'SolarWinds' TFTP server sees nothing. - Figured as much as the phone's IP
 and my TFTP server's IP are not in the same 'net.
 
 Bottom line. - I've got to figure out a way to build a 'mini-network' so the
 phone'll be happy but also set up a PC  TFTP with the same address that's
 set in the phone. - Perhaps I can fake-out the phone into thinking it's
 hitting a TFTP box on the Internet ().

Set your TFTP server's IP address to be the phone's gateway address (i.e.
192.168.0.1), and add a loopback address on your TFTP server of
62.120.xxx.xxx. Then outbound packets will hit your machine, which will
believe that the address being contacted is itself, and it will answer.

I don't know how to do that with XP. Perhaps put in another NIC and
configure it as 62.120.xxx.xxx (and plug it into a hub so that it thinks the
cable is active)

Under Linux or FreeBSD you can use something like
  ifconfig lo:0 62.120.xxx.xxx/32
  ifconfig lo alias 62.120.xxx.xxx/32
respectively.

Maybe you can boot from an Ubuntu CD (this gives you a live Linux desktop
without reinstalling your system) and set up a tftp server there.

Debug using tcpdump (Linux/FreeBSD) or ethereal (Windows)

HTH,

Brian.
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[asterisk-users] Alcatel OXO Sip

2006-09-19 Thread Christian Gatti

Hi,

Asterisk gives me an WARNING if I try to register with my alcatel oxo pbx.
Sep 18 11:27:19 WARNING[11977]: chan_sip.c:5259 check_via: Don't know how
to respond via 'SIP/2.0/udp'

An INVITE to asterisk seems to go through (debug entries in *) but the the
pbx seems to get no SIP responses.

Thanks,
Christian


 [EMAIL PROTECTED] for seqno 1663486441 (Non-critical Response)
Sep 18 11:27:19 WARNING[11977]: chan_sip.c:5259 check_via: Don't know how to
respond via 'SIP/2.0/udp'
Sep 18 11:27:20 WARNING[11977]: chan_sip.c:694 retrans_pkt: Maximum retries
exceeded on call c442405144539d17679ad928b8ec
[EMAIL PROTECTED] for seqno 1663486441 (Non-critical Response)
Sep 18 11:27:23 WARNING[11977]: chan_sip.c:5259 check_via: Don't know how to
respond via 'SIP/2.0/udp'
Sep 18 11:27:24 WARNING[11977]: chan_sip.c:694 retrans_pkt: Maximum retries
exceeded on call c442405144539d17679ad928b8ec
[EMAIL PROTECTED] for seqno 1663486441 (Non-critical Response)
 







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Re: [asterisk-users] How to Dial a number with Sangoma PRI card?

2006-09-19 Thread Mario

That's ok if I want to dial through a group. But, for my specific 
requirements, I need to dial through a specific channel. I even need to 
use the ChanIsAvail application to discover which channels are available.


Thus, without using a group, which is the correct way to dial through a PRI?


Lacy Moore - Aspendora wrote:

I dial using groups.  Dial(Zap/g1/1234)
 
I'm pretty sure this was taken off of the examples on the Sangoma website.
 
 
On 9/19/06, *Mario* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


I have a Sangoma PRI card configured for E1 line (i.e. 30+1
channels per
port) and I'm not quite sure on how the Dial command should
performed.

I'm using the standard Dial command as if it were a Zap channel.
For example

   Dial(Zap/1/1234) ; Dial 1234 using channel 1 on port 1
   Dial(Zap/32/1234) ; Dial 1234 using channel 1 on port 2 (i.e. 32nd
zapata entry)

Most often than not this works, but sometimes the call fails. However,
reading the Asterisk docs, it says that to dial using a PRI card I
should use, instead, the following command:

   Dial(Zap/1-1/1234) ; Dial using channel 1 on port 1
   Dial(Zap/2-1/1234) ; Dial using channel 1 on port 2

Which one is correct? Should I dial Zap/32 or Zap/2-1 to go
through the
1st channel on port 2?

I'm scared of changing my whole dial plan and then discover that,
occasionally, things do not work as expected.

Please, can someone who has used Sangoma PRI card help me? My
Zapata.conf is set as if we had 60+ channels (something similar to
this):

   context = my_context
   group = 1
   [snip...]
   signalling = pri_cpe
   switchtype = euroisdn
   channel = 1-15, 17-31 ; Same, up to channel 62

Thanks in advance.
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--
Lacy Moore
Aspendora, Inc. 


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Re: [asterisk-users] Accounting and re-invite

2006-09-19 Thread Simon Woodhead

Hi Ronald,On 9/19/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
I am thinking if re-invite will interfere accounting.No it won't 
Please help me to figure it out:Phone A is registered at asterisk and calls a gateway. If the gatewayallows re-invite than the rtp would go directly from phone A to thegateway, while the sip messages are still going through Asterisk.
Asterisk will be informed when the call ended.If it is a postpaid accounting, just bill the customer, however, how isit for a pre-paid (calling card user)?I think Asterisk will have no power to turn off the call from A to the
gateway.Even more, if the gateway would allow to end a call and continue with anew call, the new call would not be billed (or would it)?The SIP messages control the call, irrespective of where the RTP goes so Asterisk can terminate/set-up calls exactly as if the RTP was being handled. 
I guess the solution must be re-invite=noHowever, re-invite=no means that each call is going with rtp also
through my server, what means for a remote phone, I have to provide forboth legs the bandwidth.Personally, I would always handle the RTP on quality/accountability/consitency grounds but, yes, that will incur a bandwidth overhead.

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Re: [asterisk-users] Enabling Second Processor Trashes Audio Quality

2006-09-19 Thread Tzafrir Cohen

On Mon, Sep 18, 2006 at 05:07:31PM -0700, George Pajari wrote:
 Any thoughts on this one?
 
 IBM xSeries 330 processor running Debian 3.1 (2.6.8 kernel) with a 
 TE406P board.
 
 Working fine (more or less) connected to a couple of PRIs.
 
 Rebuild kernel with support for second CPU and inbound (PRI - SIP) 
 audio is badly garbled. Outbound (Asterisk - PRI) is fine.

BTW: If you're using the provided kernel, why not use the provided -smp
packages?

 
 Rebooting a kernel with support for only a single CPU clears up the problem

Is it the original kernel or a boot option to disable SMP?

 
 There is a small possibility that the TE406P card is acting up and that 
 the audio problem is coincidental with the switch between 
 dual-processor/single-processor kernels but thought I'd consult the list 
 for advice.
 
 Will be swapping out the TE406P for a new TE407P in the next couple of 
 days and will report findings then.
 
 g.
 
 -- 
 George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
 Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
 Hosted IP PBX Services for SOHO  Small Businesses - www.ip-centrex.ca
 VoIP Service, Equipment, Systems, and Consulting - www.netvoice.ca

minor, off-topic, comment: how can I call you through VOIP ?

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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RE: [asterisk-users] Alcatel OXO Sip

I think that (one of the) offending line(s) is in chan_sip.c:

if (strcasecmp(via, SIP/2.0/UDP)) {
ast_log(LOG_WARNING, Don't know how to respond
via '%s'\n, via);
return -1;
}

This is looking for an upper-case 'UDP' whereas your oxo pbx is using a
lower-case 'udp'.
 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Christian Gatti
 Sent: 19 September 2006 10:08
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Alcatel OXO Sip
 
 Hi,
 
 Asterisk gives me an WARNING if I try to register with my 
 alcatel oxo pbx.
 Sep 18 11:27:19 WARNING[11977]: chan_sip.c:5259 check_via: 
 Don't know 
 how
 to respond via 'SIP/2.0/udp'
 
 An INVITE to asterisk seems to go through (debug entries in 
 *) but the the pbx seems to get no SIP responses.
 
 Thanks,
 Christian
 
 
  [EMAIL PROTECTED] for seqno 1663486441 (Non-critical 
 Response) Sep 18 11:27:19 WARNING[11977]: chan_sip.c:5259 
 check_via: Don't know how to respond via 'SIP/2.0/udp'
 Sep 18 11:27:20 WARNING[11977]: chan_sip.c:694 retrans_pkt: 
 Maximum retries exceeded on call c442405144539d17679ad928b8ec
 [EMAIL PROTECTED] for seqno 1663486441 (Non-critical 
 Response) Sep 18 11:27:23 WARNING[11977]: chan_sip.c:5259 
 check_via: Don't know how to respond via 'SIP/2.0/udp'
 Sep 18 11:27:24 WARNING[11977]: chan_sip.c:694 retrans_pkt: 
 Maximum retries exceeded on call c442405144539d17679ad928b8ec
 [EMAIL PROTECTED] for seqno 1663486441 (Non-critical Response)
  
 
 
 
 
 
 
 
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RE: [asterisk-users] Alcatel OXO Sip

Oh, when I said offending line I didn't mean to imply that Asterisk is
wrong - I think that the OXO PBX should be using upper-case. Sorry. 

 -Original Message-
 From: Steve Langstaff 
 Sent: 19 September 2006 10:18
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [asterisk-users] Alcatel OXO Sip
 
 I think that (one of the) offending line(s) is in chan_sip.c:
 
   if (strcasecmp(via, SIP/2.0/UDP)) {
   ast_log(LOG_WARNING, Don't know how to 
 respond via '%s'\n, via);
   return -1;
   }
 
 This is looking for an upper-case 'UDP' whereas your oxo pbx 
 is using a lower-case 'udp'.
  
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  Christian Gatti
  Sent: 19 September 2006 10:08
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] Alcatel OXO Sip
  
  Hi,
  
  Asterisk gives me an WARNING if I try to register with my 
 alcatel oxo 
  pbx.
  Sep 18 11:27:19 WARNING[11977]: chan_sip.c:5259 check_via: 
  Don't know
  how
  to respond via 'SIP/2.0/udp'
  
  An INVITE to asterisk seems to go through (debug entries in
  *) but the the pbx seems to get no SIP responses.
  
  Thanks,
  Christian
  
  
   [EMAIL PROTECTED] for seqno 1663486441 (Non-critical
  Response) Sep 18 11:27:19 WARNING[11977]: chan_sip.c:5259
  check_via: Don't know how to respond via 'SIP/2.0/udp'
  Sep 18 11:27:20 WARNING[11977]: chan_sip.c:694 retrans_pkt: 
  Maximum retries exceeded on call c442405144539d17679ad928b8ec
  [EMAIL PROTECTED] for seqno 1663486441 (Non-critical
  Response) Sep 18 11:27:23 WARNING[11977]: chan_sip.c:5259
  check_via: Don't know how to respond via 'SIP/2.0/udp'
  Sep 18 11:27:24 WARNING[11977]: chan_sip.c:694 retrans_pkt: 
  Maximum retries exceeded on call c442405144539d17679ad928b8ec
  [EMAIL PROTECTED] for seqno 1663486441 (Non-critical Response)
   
  
  
  
  
  
  
  
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Re: [asterisk-users] How to Dial a number with Sangoma PRI card?

2006-09-19 Thread yusuf

hi,

I did it like this:
I wrote a PHP AGI script, that I call from the dial plan.
In the AGI I check

fwrite(STDOUT,CHANNEL STATUS $currchan \n);
fflush(STDOUT);

where currchan is ZAP/1 for instance. It returns whether the channel is used. I then pass this
back as a variable back to dial plan, and I use that variable to dial.

HTH

Mario wrote:
That's ok if I want to dial through a group. But, for my specific
requirements, I need to dial through a specific channel. I even need to
use the ChanIsAvail application to discover which channels are available.

Thus, without using a group, which is the correct way to dial through a
PRI?

Lacy Moore - Aspendora wrote:

I dial using groups. Dial(Zap/g1/1234)

I'm pretty sure this was taken off of the examples on the Sangoma
website.

On 9/19/06, *Mario* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

I have a Sangoma PRI card configured for E1 line (i.e. 30+1
channels per
port) and I'm not quite sure on how the Dial command should
performed.

I'm using the standard Dial command as if it were a Zap channel.
For example

Dial(Zap/1/1234) ; Dial 1234 using channel 1 on port 1
Dial(Zap/32/1234) ; Dial 1234 using channel 1 on port 2 (i.e. 32nd
zapata entry)

Most often than not this works, but sometimes the call fails.
However,

reading the Asterisk docs, it says that to dial using a PRI card I
should use, instead, the following command:

Dial(Zap/1-1/1234) ; Dial using channel 1 on port 1
Dial(Zap/2-1/1234) ; Dial using channel 1 on port 2

Which one is correct? Should I dial Zap/32 or Zap/2-1 to go
through the
1st channel on port 2?

I'm scared of changing my whole dial plan and then discover that,
occasionally, things do not work as expected.

Please, can someone who has used Sangoma PRI card help me? My
Zapata.conf is set as if we had 60+ channels (something similar to
this):

context = my_context
group = 1
[snip...]
signalling = pri_cpe
switchtype = euroisdn
channel = 1-15, 17-31 ; Same, up to channel 62

Thanks in advance.
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--
Lacy Moore
Aspendora, Inc.

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--
thanks,
yusuf

--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

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RE: [asterisk-users] Alcatel OXO Sip

2006-09-19 Thread Patrick

On Tue, 2006-09-19 at 02:18 -0700, Steve Langstaff wrote:
 I think that (one of the) offending line(s) is in chan_sip.c:
 
   if (strcasecmp(via, SIP/2.0/UDP)) {
   ast_log(LOG_WARNING, Don't know how to respond
 via '%s'\n, via);
   return -1;
   }
 
 This is looking for an upper-case 'UDP' whereas your oxo pbx is using a
 lower-case 'udp'.

man strcasecmp says:
strcasecmp, strncasecmp - compare two strings ignoring case

So the case is ignored and either sip and SIP or udp and UDP should
work.

Regards,
Patrick


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Re: [asterisk-users] How to Dial a number with Sangoma PRI card?

2006-09-19 Thread Mario


ysuf,

that's exactly what I'm doing (in Python instead of PHP, but that 
doesn't matter). However, my question is: should I ask if ZAP/1 is 
available or if  ZAP/1-1 is available? For example:


   ChanIsAvail(Zap/1Zap/2Zap/3)

or

   ChanIsAvail(Zap/1-1Zap/1-2Zap/1-3)

And, once discovered which channel is available, which form of Dial 
should I use? Should I say:


   Dial(Zap/2/1234)

or

   Dial(Zap/1-2/1234)

yusuf wrote:

hi,

I did it like this:
I wrote a PHP AGI script, that I call from the dial plan.
In the AGI I check

 fwrite(STDOUT,CHANNEL STATUS $currchan \n);
fflush(STDOUT);

where currchan is ZAP/1 for instance.  It returns whether the channel 
is used.  I then pass this back as a variable back to dial plan, and I 
use that variable to dial.


HTH

Mario wrote:
That's ok if I want to dial through a group. But, for my specific 
requirements, I need to dial through a specific channel. I even need 
to use the ChanIsAvail application to discover which channels are 
available.


Thus, without using a group, which is the correct way to dial through 
a PRI?



Lacy Moore - Aspendora wrote:


I dial using groups.  Dial(Zap/g1/1234)
 
I'm pretty sure this was taken off of the examples on the Sangoma 
website.
 
 
On 9/19/06, *Mario* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


I have a Sangoma PRI card configured for E1 line (i.e. 30+1
channels per
port) and I'm not quite sure on how the Dial command should
performed.

I'm using the standard Dial command as if it were a Zap channel.
For example

   Dial(Zap/1/1234) ; Dial 1234 using channel 1 on port 1
   Dial(Zap/32/1234) ; Dial 1234 using channel 1 on port 2 (i.e. 
32nd

zapata entry)

Most often than not this works, but sometimes the call fails. 
However,

reading the Asterisk docs, it says that to dial using a PRI card I
should use, instead, the following command:

   Dial(Zap/1-1/1234) ; Dial using channel 1 on port 1
   Dial(Zap/2-1/1234) ; Dial using channel 1 on port 2

Which one is correct? Should I dial Zap/32 or Zap/2-1 to go
through the
1st channel on port 2?

I'm scared of changing my whole dial plan and then discover that,
occasionally, things do not work as expected.

Please, can someone who has used Sangoma PRI card help me? My
Zapata.conf is set as if we had 60+ channels (something similar to
this):

   context = my_context
   group = 1
   [snip...]
   signalling = pri_cpe
   switchtype = euroisdn
   channel = 1-15, 17-31 ; Same, up to channel 62

Thanks in advance.
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--
Lacy Moore
Aspendora, Inc. 



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Re: [asterisk-users] ANI and Meetme...

Natambu Obleton wrote:
 
 
 Ok. First question is how to make it say my number back.
 
 Like if you call extension 1000 from extension 1001, I want it to say
 “Number is 1,0,0,1” like an ANI number? Help.
 
  
 
  
 
 Also I want to setup a meetme conference so that it asks “Enter
 conference number” then execute meetme($entered_number)
 
  
 
  
 
 I feel dumb asking because these sound like they should be so easy, but
 I can’t find any help with this. Thanks.
 
  
 
  
 
 Natambu Obleton
 
 Network Engineer
 
 FastTrack Communications
 
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 
 (970) 247-3366 office
 
 (970) 247-2426 fax
 
  
 
  
 
 
 
 
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The first item (repeating ANI number):

; Use the saydigits app to repeat the ANI to the caller
exten = _X.,1,Answer()
exten = _X.,n,Wait(2)
exten = _X.,n,SayDigits(CALLERID(ani))

Cheers
-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] When does Scalability requests Asterisk to Use SER ?

Marco Mouta wrote:
 Hi all,
 
 I'm planing to develop a solution based on Asterisk for about 300 users.
 My question now is, do I really need to use openSER as the sip proxy and
 Asterisk for the PBX functions?
 
 Can i trust in a solution only with Asterisk to make all this install?
 
 Please help me with your experience on this kind of asterisk solutions.
 
 I've googled and read about asterisk at large scale solutions, but still
 in doubt.
 http://www.voip-info.org/wiki-Asterisk+at+large
 
 
 -- 
 Com os melhores cumprimentos,
 
 Marco Mouta
 
 
 
 
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In my experience, yes you can use *just* asterisk for the implementation
of a large scale setup, you just better be sure you've planned it out
well. I've set up a few large scale Asterisk implementations, covering
more than 1K users on a single box. And that was in 2005 using trunk.
There were problems, but all in all it was (and is, for the former
client) not a bad implementation. If you're just looking at a large PBX
install, you're definitely fine with a well planned system.

Just my $0.02, not to be taken as a guarantee ;-)


-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] How to Dial a number with Sangoma PRI card?

2006-09-19 Thread yusuf


Mario,

try ChanIsAvail(Zap/1-1)

but when you dial, its Zap/1/${EXTEN}

HTH

Mario wrote:

ysuf,

that's exactly what I'm doing (in Python instead of PHP, but that 
doesn't matter). However, my question is: should I ask if ZAP/1 is 
available or if  ZAP/1-1 is available? For example:


   ChanIsAvail(Zap/1Zap/2Zap/3)

or

   ChanIsAvail(Zap/1-1Zap/1-2Zap/1-3)

And, once discovered which channel is available, which form of Dial 
should I use? Should I say:


   Dial(Zap/2/1234)

or

   Dial(Zap/1-2/1234)

yusuf wrote:


hi,

I did it like this:
I wrote a PHP AGI script, that I call from the dial plan.
In the AGI I check

 fwrite(STDOUT,CHANNEL STATUS $currchan \n);
fflush(STDOUT);

where currchan is ZAP/1 for instance.  It returns whether the channel 
is used.  I then pass this back as a variable back to dial plan, and I 
use that variable to dial.


HTH

Mario wrote:

That's ok if I want to dial through a group. But, for my specific 
requirements, I need to dial through a specific channel. I even need 
to use the ChanIsAvail application to discover which channels are 
available.


Thus, without using a group, which is the correct way to dial through 
a PRI?



Lacy Moore - Aspendora wrote:


I dial using groups.  Dial(Zap/g1/1234)
 
I'm pretty sure this was taken off of the examples on the Sangoma 
website.
 
 
On 9/19/06, *Mario* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


I have a Sangoma PRI card configured for E1 line (i.e. 30+1
channels per
port) and I'm not quite sure on how the Dial command should
performed.

I'm using the standard Dial command as if it were a Zap channel.
For example

   Dial(Zap/1/1234) ; Dial 1234 using channel 1 on port 1
   Dial(Zap/32/1234) ; Dial 1234 using channel 1 on port 2 (i.e. 
32nd

zapata entry)

Most often than not this works, but sometimes the call fails. 
However,

reading the Asterisk docs, it says that to dial using a PRI card I
should use, instead, the following command:

   Dial(Zap/1-1/1234) ; Dial using channel 1 on port 1
   Dial(Zap/2-1/1234) ; Dial using channel 1 on port 2

Which one is correct? Should I dial Zap/32 or Zap/2-1 to go
through the
1st channel on port 2?

I'm scared of changing my whole dial plan and then discover that,
occasionally, things do not work as expected.

Please, can someone who has used Sangoma PRI card help me? My
Zapata.conf is set as if we had 60+ channels (something similar to
this):

   context = my_context
   group = 1
   [snip...]
   signalling = pri_cpe
   switchtype = euroisdn
   channel = 1-15, 17-31 ; Same, up to channel 62





--
thanks,
yusuf

--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

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Re[2]: [asterisk-users] How to Dial a number with Sangoma PRI card?

2006-09-19 Thread Melcon Moraes

I just use Dial(Zap/1/1234)

[]'s
MM

 -Original Message-
From:   Mario [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Cc: 
Sent:  Tue, 19 Sep 2006 12:17:07 +0200
Delivered:  Tue,  19 Sep 2006 07:06:25 
Subject:[asterisk-users] How to Dial a number with Sangoma PRI card?

ysuf,

that's exactly what I'm doing (in Python instead of PHP, but that 
doesn't matter). However, my question is: should I ask if ZAP/1 is 
available or if  ZAP/1-1 is available? For example:

ChanIsAvail(Zap/1Zap/2Zap/3)

or

ChanIsAvail(Zap/1-1Zap/1-2Zap/1-3)

And, once discovered which channel is available, which form of Dial 
should I use? Should I say:

Dial(Zap/2/1234)

or

Dial(Zap/1-2/1234)

yusuf wrote:
 hi,

 I did it like this:
 I wrote a PHP AGI script, that I call from the dial plan.
 In the AGI I check

  fwrite(STDOUT,CHANNEL STATUS $currchan \n);
 fflush(STDOUT);

 where currchan is ZAP/1 for instance.  It returns whether the channel 
 is used.  I then pass this back as a variable back to dial plan, and I 
 use that variable to dial.

 HTH

 Mario wrote:
 That's ok if I want to dial through a group. But, for my specific 
 requirements, I need to dial through a specific channel. I even need 
 to use the ChanIsAvail application to discover which channels are 
 available.

 Thus, without using a group, which is the correct way to dial through 
 a PRI?

 Lacy Moore - Aspendora wrote:

 I dial using groups.  Dial(Zap/g1/1234)

 I'm pretty sure this was taken off of the examples on the Sangoma 
 website.

 On 9/19/06, *Mario* [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED] wrote:

 I have a Sangoma PRI card configured for E1 line (i.e. 30+1
 channels per
 port) and I'm not quite sure on how the Dial command should
 performed.

 I'm using the standard Dial command as if it were a Zap channel.
 For example

Dial(Zap/1/1234) ; Dial 1234 using channel 1 on port 1
Dial(Zap/32/1234) ; Dial 1234 using channel 1 on port 2 (i.e. 
 32nd
 zapata entry)

 Most often than not this works, but sometimes the call fails. 
 However,
 reading the Asterisk docs, it says that to dial using a PRI card I
 should use, instead, the following command:

Dial(Zap/1-1/1234) ; Dial using channel 1 on port 1
Dial(Zap/2-1/1234) ; Dial using channel 1 on port 2

 Which one is correct? Should I dial Zap/32 or Zap/2-1 to go
 through the
 1st channel on port 2?

 I'm scared of changing my whole dial plan and then discover that,
 occasionally, things do not work as expected.

 Please, can someone who has used Sangoma PRI card help me? My
 Zapata.conf is set as if we had 60+ channels (something similar to
 this):

context = my_context
group = 1
[snip...]
signalling = pri_cpe
switchtype = euroisdn
channel = 1-15, 17-31 ; Same, up to channel 62

 Thanks in advance.
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 -- 
 Lacy Moore
 Aspendora, Inc. 

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E-mail classificado pelo Identificador de Spam Inteligente Terra.
Para alterar a categoria classificada, visite
http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1158661729.120148.30995.curepipe.hst.terra.com.br,6932,Des15,Des15

 --Original Message Ends--

-- 
Melcon Moraes [EMAIL PROTECTED]

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Re: [asterisk-users] How to Dial a number with Sangoma PRI card?

2006-09-19 Thread Tristan


You should try :

exten = _,n,ChanIsAvail(Zap/XY)
exten = _,n,NoOp(AvailChannel=${AVAILCHAN})
exten = _,n,Set(__DialChannel=${CUT(AVAILCHAN,,1)})
exten = _,n,Dial(${DialChannel}/YOURNUMTODIAL)


Where X stands for the strategy to fill your PRI ( r,R,g,G,.. ) and Y 
stands for the trunk group to use.


${AVAILCHAN} will contain the name of the next available channel, for 
example Zap/32-1.


You cut the -1 with CUT then you can use it in dial app...

Regards,

Tristan Mahé

Mario a écrit :

ysuf,

that's exactly what I'm doing (in Python instead of PHP, but that 
doesn't matter). However, my question is: should I ask if ZAP/1 is 
available or if  ZAP/1-1 is available? For example:


   ChanIsAvail(Zap/1Zap/2Zap/3)

or

   ChanIsAvail(Zap/1-1Zap/1-2Zap/1-3)

And, once discovered which channel is available, which form of Dial 
should I use? Should I say:


   Dial(Zap/2/1234)

or

   Dial(Zap/1-2/1234)

yusuf wrote:

hi,

I did it like this:
I wrote a PHP AGI script, that I call from the dial plan.
In the AGI I check

 fwrite(STDOUT,CHANNEL STATUS $currchan \n);
fflush(STDOUT);

where currchan is ZAP/1 for instance.  It returns whether the channel 
is used.  I then pass this back as a variable back to dial plan, and 
I use that variable to dial.


HTH

Mario wrote:
That's ok if I want to dial through a group. But, for my specific 
requirements, I need to dial through a specific channel. I even need 
to use the ChanIsAvail application to discover which channels are 
available.


Thus, without using a group, which is the correct way to dial 
through a PRI?



Lacy Moore - Aspendora wrote:


I dial using groups.  Dial(Zap/g1/1234)
 
I'm pretty sure this was taken off of the examples on the Sangoma 
website.
 
 
On 9/19/06, *Mario* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


I have a Sangoma PRI card configured for E1 line (i.e. 30+1
channels per
port) and I'm not quite sure on how the Dial command should
performed.

I'm using the standard Dial command as if it were a Zap channel.
For example

   Dial(Zap/1/1234) ; Dial 1234 using channel 1 on port 1
   Dial(Zap/32/1234) ; Dial 1234 using channel 1 on port 2 
(i.e. 32nd

zapata entry)

Most often than not this works, but sometimes the call fails. 
However,

reading the Asterisk docs, it says that to dial using a PRI card I
should use, instead, the following command:

   Dial(Zap/1-1/1234) ; Dial using channel 1 on port 1
   Dial(Zap/2-1/1234) ; Dial using channel 1 on port 2

Which one is correct? Should I dial Zap/32 or Zap/2-1 to go
through the
1st channel on port 2?

I'm scared of changing my whole dial plan and then discover that,
occasionally, things do not work as expected.

Please, can someone who has used Sangoma PRI card help me? My
Zapata.conf is set as if we had 60+ channels (something similar to
this):

   context = my_context
   group = 1
   [snip...]
   signalling = pri_cpe
   switchtype = euroisdn
   channel = 1-15, 17-31 ; Same, up to channel 62

Thanks in advance.
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--
Lacy Moore
Aspendora, Inc. 



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[asterisk-users] iax2 trunk call limits

2006-09-19 Thread Ma Zhiyong




Hi, all 

Can I limit calls in one iax2 trunk just like sip peers do? 
How?


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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-19 Thread Benjamin Jacob


Rushowr wrote:


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Date:
Tue, 19 Sep 2006 05:52:52 -0500




Marco Mouta wrote:
 


Hi all,

I'm planing to develop a solution based on Asterisk for about 300 users.
My question now is, do I really need to use openSER as the sip proxy and
Asterisk for the PBX functions?

Can i trust in a solution only with Asterisk to make all this install?

Please help me with your experience on this kind of asterisk solutions.

I've googled and read about asterisk at large scale solutions, but still
in doubt.
http://www.voip-info.org/wiki-Asterisk+at+large


--
Com os melhores cumprimentos,

Marco Mouta




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In my experience, yes you can use *just* asterisk for the implementation
of a large scale setup, you just better be sure you've planned it out
well. I've set up a few large scale Asterisk implementations, covering
more than 1K users on a single box. And that was in 2005 using trunk.
There were problems, but all in all it was (and is, for the former
client) not a bad implementation. If you're just looking at a large PBX
install, you're definitely fine with a well planned system.

Just my $0.02, not to be taken as a guarantee ;-)


 

You mean, 1 K simultaneous calls? or 1 K registered users(yes yes. this 
is the one!!)???


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[asterisk-users] Anyone Using a Patton (Inalp) SmartNode 2400 for T.38?

2006-09-19 Thread George Pajari

We're having fun trying to get a Patton (Inalp) SmartNode 2400 to 
function as a T.38/PRI gateway with Asterisk handling the pass-through.


Any other SN2400 users out there with forehead-shaped dents in their walls?

--
George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
Hosted IP PBX Services for SOHO  Small Businesses - www.ip-centrex.ca
VoIP Service, Equipment, Systems, and Consulting - www.netvoice.ca

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Re: [asterisk-users] rxfax, spandsp and lack of ecm

2006-09-19 Thread Artifex Maximus


I had changed my setup to iaxmodem/hylafax and that is wonderful.
Works like charm. Thanks your help!

I have problem with printing incoming faxes because it's likely skip
the header and the footer on some fax. I receive the fax on email as
well and pdf is perfect so there is some problem (might related to
resizing) in fax2ps/lp combo.

bye,
Zsolt

On 9/18/06, Artifex Maximus [EMAIL PROTECTED] wrote:

I'm using snapshot 20060915 for days and it's much better than before.
Still have some missing lines might related to bad quality line.

Thanks again!

bye,
Zsolt

On 9/13/06, Steve Underwood [EMAIL PROTECTED] wrote:
 Hi Bruce,

 Looks like your typing is as bad as mine :-)

 Try http://www.soft-switch.org/downloads/snapshots/spandsp

 Steve

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Re: [asterisk-users] Digium GUI?

2006-09-19 Thread Steve Totaro

Someone already said that they saw it at VON.  It was super simple to 
change the look and branding but the UI itself was nothing too special.


Thanks,
Steve

Douglas Garstang wrote:

I wonder if the look and feel of this GUI will be completely configurable. If 
it's not, then I really don't think that's very useful. Service providers 
wouldn't be able to use it to let their customers manage their own settings, 
and customers wouldn't want to use it if it wasn't branded with their company 
info.
 


\


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[asterisk-users] Query ,NEED help regarding MWI

2006-09-19 Thread Tanzeel serfaraz

Hi Users;
i have to implement MWI scenario like this:

IPphone,ATAopenserAsterisk

my users are registered at openser and voicemail box
is configured at asterisk.

MWI is send by ASTERISK to OPENSER and then OPENSER to
IPPHONE OR ATA.

My query is this;

Q:let say i got a NOTIFY message from openser to
IPPHONE/ATA and my  phone changes the tone to a MWI
tone,  Now if the Telephone adapter got reset by any
reason , the MWI tone gets lost ,so HOW CAN I FIX THIS
PROBLEM?
 
I mean how openser know that phone got disconnected so
that it again send NOTIFY message to IPPHONE or is
there any way to send NOTIFY message peridically after
a define time stamp.

Hope someone would help me
Thanks  





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Re: [asterisk-users] iax2 trunk call limits

2006-09-19 Thread Doug Lytle


Ma Zhiyong wrote:

Hi, all
Can I limit calls in one iax2 trunk just like sip peers do? How?


http://www.voip-info.org/wiki/view/Asterisk+cmd+CheckGroup

Doug


--

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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Re: [asterisk-users] rxfax, spandsp and lack of ecm

2006-09-19 Thread Doug Lytle


Artifex Maximus wrote:

I have problem with printing incoming faxes because it's likely skip
the header and the footer on some fax. I receive the fax on email as


I would suggest asking this question on the HylaFAX mailing list.

Doug


--

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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RE: [asterisk-users] Alcatel OXO Sip

Oops! Yes, I was confusing strcmp and strcasecmp, sorry - too little
coffee.

However, it's funny that the strcasecmp of SIP/2.0/UDP fails to match
with SIP/2.0/udp. Almost like it's using a strcmp instead of
strcasecmp!

 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Christian Gatti

Asterisk gives me an WARNING if I try to register with my alcatel oxo
pbx.
Sep 18 11:27:19 WARNING[11977]: chan_sip.c:5259 check_via: Don't know 
how to respond via 'SIP/2.0/udp' 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Patrick
 Sent: 19 September 2006 10:57
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] Alcatel OXO Sip
 
 On Tue, 2006-09-19 at 02:18 -0700, Steve Langstaff wrote:
  I think that (one of the) offending line(s) is in chan_sip.c:
  
  if (strcasecmp(via, SIP/2.0/UDP)) {
  ast_log(LOG_WARNING, Don't know how to 
 respond via '%s'\n, via);
  return -1;
  }
  
  This is looking for an upper-case 'UDP' whereas your oxo 
 pbx is using 
  a lower-case 'udp'.
 
 man strcasecmp says:
 strcasecmp, strncasecmp - compare two strings ignoring case
 
 So the case is ignored and either sip and SIP or udp and UDP 
 should work.
 
 Regards,
 Patrick
 
 
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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

Benjamin Jacob wrote:
 Rushowr wrote:
 
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 Date:
 Tue, 19 Sep 2006 05:52:52 -0500


 

 Marco Mouta wrote:
  

 Hi all,

 I'm planing to develop a solution based on Asterisk for about 300 users.
 My question now is, do I really need to use openSER as the sip proxy and
 Asterisk for the PBX functions?

 Can i trust in a solution only with Asterisk to make all this install?

 Please help me with your experience on this kind of asterisk solutions.

 I've googled and read about asterisk at large scale solutions, but still
 in doubt.
 http://www.voip-info.org/wiki-Asterisk+at+large


 -- 
 Com os melhores cumprimentos,

 Marco Mouta


 

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 In my experience, yes you can use *just* asterisk for the implementation
 of a large scale setup, you just better be sure you've planned it out
 well. I've set up a few large scale Asterisk implementations, covering
 more than 1K users on a single box. And that was in 2005 using trunk.
 There were problems, but all in all it was (and is, for the former
 client) not a bad implementation. If you're just looking at a large PBX
 install, you're definitely fine with a well planned system.

 Just my $0.02, not to be taken as a guarantee ;-)


  

 You mean, 1 K simultaneous calls? or 1 K registered users(yes yes. this
 is the one!!)???
 
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Sorry, should have been a little more specific. I've had Asterisk
running realtime SIP users/peers and realtime sql calls from the
dialplan (all with MySQL), and have had around 2.5k registered users and
a peak (that I recall) of around 500 concurrent calls.

-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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[asterisk-users] Clustering architecture and echo cancellation issue

2006-09-19 Thread Panagiotis Zikos

Hi,I want to setup a VOIP call center that will be also able to send calls to PSTN over TE110P or TE205P.The first question is if i need to go with a clustering architecture (meaning that i am going to need two PCs and two cards) or a single (strong) PC with one card is sufficient?Secondly the TE205P card supports echo cancellation on board. Does this mean that the other one will expose echo problems? Can i overcome them?Thanks alot 
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[asterisk-users] Aastra 9133i and Atcom AT-320 - Comments please

2006-09-19 Thread James Dyer

I'm planning to deploy an Asterisk system in our office soon, and am 
thinking of using a mixture of Aastra 9133i and Atcom AT-320 phones.

Has anyone got any comments (good or bad) about these phone models?

Thanks,

James

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Re: [asterisk-users] Aastra 9133i and Atcom AT-320 - Comments please

2006-09-19 Thread bails

I've never used the Aastra but the AT-320's seem to work fairly well, my 
only bug with them is there lack of weight, they slide across the desk 
to readily.


Bails


James Dyer wrote:
I'm planning to deploy an Asterisk system in our office soon, and am 
thinking of using a mixture of Aastra 9133i and Atcom AT-320 phones.


Has anyone got any comments (good or bad) about these phone models?

Thanks,

James

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[asterisk-users] Re: Calls on hold

2006-09-19 Thread Mir


Doesnt anyone know if this is possible?


2006/9/13, Mir [EMAIL PROTECTED]:

Hello

Is there a possibility for sending an event on the managerinterface
(AMI) when a call is put on/off hold?

Or is there any other way to detect when a call is placed on hold?

Michael


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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-19 Thread burke

 Benjamin Jacob wrote:
 Rushowr wrote:

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 Date:
 Tue, 19 Sep 2006 05:52:52 -0500


 

 Marco Mouta wrote:


 Hi all,

 I'm planing to develop a solution based on Asterisk for about 300
 users.
 My question now is, do I really need to use openSER as the sip proxy
 and
 Asterisk for the PBX functions?

 Can i trust in a solution only with Asterisk to make all this install?

 Please help me with your experience on this kind of asterisk
 solutions.

 I've googled and read about asterisk at large scale solutions, but
 still
 in doubt.
 http://www.voip-info.org/wiki-Asterisk+at+large


 --
 Com os melhores cumprimentos,

 Marco Mouta


 

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 In my experience, yes you can use *just* asterisk for the
 implementation
 of a large scale setup, you just better be sure you've planned it out
 well. I've set up a few large scale Asterisk implementations, covering
 more than 1K users on a single box. And that was in 2005 using trunk.
 There were problems, but all in all it was (and is, for the former
 client) not a bad implementation. If you're just looking at a large PBX
 install, you're definitely fine with a well planned system.

 Just my $0.02, not to be taken as a guarantee ;-)




 You mean, 1 K simultaneous calls? or 1 K registered users(yes yes. this
 is the one!!)???

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 Sorry, should have been a little more specific. I've had Asterisk
 running realtime SIP users/peers and realtime sql calls from the
 dialplan (all with MySQL), and have had around 2.5k registered users and
 a peak (that I recall) of around 500 concurrent calls.

 --
 S McGowan
 VoIP Consultant
 [EMAIL PROTECTED]


Can you explain your design in a little more detail? What kind of hardware
did you use to get over 1k users on a single box and 500 concurrent calls?
Sounds like a very interesting medium-large scale implementation that
others could learn from.

thanks,
Ryan
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Re: [asterisk-users] How to install HUDLite Server

2006-09-19 Thread Nicolás Gudiño


And also it would be nice to have more detailed instructions on how to
upgrade FOP.


This belongs to FOP mailing list... but anyways:

0) backup your previous install, just in case
1) replace op_server.pl
2) replace operator_panel.swf
3) read UPGRADE, you might need to add 3 or 4 new parameters to
op_style.cfg and maybe op_server.cfg and op_buttons.cfg. Those
parameters are usually optional, so this step is probably optional.
4) restart the panel

If you are upgrading for an old version and you are using an outdated
version of FreePBX/AMP, you might need to modify the script that
generate the config file (retrieve_op_conf_from_mysql.pl) in order to
display queue activity, because queue buttons where renamed from
[queuename] to [QUEUE/quename]. That script is part of AMP/FreePBX,
not FOP.

All of this is documented somehow in the tarball, online documentation.

You are always welcome to improve the documentation if you feel is
innacurate or incomplete, make translations, etc.

And if you do not have time to do custom mods, you can hire someone to
do it for you... ;)

Best regards,


--
Nicolás Gudiño
Buenos Aires - Argentina
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[asterisk-users] Format_MP3, Streaming, File Formats, MOH

2006-09-19 Thread Robert Chadwell









Format_MP3 appears to play MOH files starting at the
beginning of each file, using the .wav file format, making for some repetitive hold
music unless you alter the file itself to begin somewhere in the middle.



Solution: One stream that all users connect to 
giving dynamic hold music (tried and tested in A1.0x using mpg123 with some
success, and Icecast or Slimserver or Shoutcast)



Format_MP3 doesnt seem to stream, and the wiki is
wrong about streamplayer being used to play streams, as it is only used to play
raw TCP streams. 



There are many questions in forums on the
web with no answers about how to solve this dilemma, How do you get users
connected to a constantly-changing stream of music instead of streams starting
from the beginning (regardless of whether Linux counts them as one stream or
not where the processor is concerned)?



Hopefully, at the end of this thread, I will have enough
information to go back to these web-forums and post the answer. To get it
started  here is what I have tried that hasnt worked. In most all
cases the response is Music on hold started, immediately followed
by Music on hold stopped with no sound in any case.



;[classes]

;mode=custom

;application=/usr/bin/streamplayer 194.158.114.67 8000

;format=ulaw

--- Straight From The Music On Hold Wiki



;default = quietmp3:/var/lib/asterisk/mohmp3-dummy
-@,http://www.shoutcast.com/sbin/tunein-station.pls?id=7733filename=playlist.pls

--- From the Nerd Vittles Tutorial with the -@ added
because mpg123 seemed to ask for it since the file was a .pls



;default = mp3:http://127.0.0.1:9000/stream.mp3

-- From a forum of someone using mpg123 to stream
SlimServer (installed mpg123 v0.60 with no success here)



[default]

mode=files

directory=
/var/lib/asterisk/mohmp3-dummy,http://127.0.0.1:9000/stream.mp3

-- Tried a 1.2 format



;default =
quietmp3:/var/lib/asterisk/mohmp3-dummy,http://193.251.154.243:8000/

-- Thought maybe it was SlimServer  so tried
to stream the top Shoutcast station



;default = quietmp3:/var/lib/asterisk/mohmp3-dummy,http://127.0.0.1:9000/stream.mp3

-- Tried to stream Slimserver using the old format





Thank you in advance  I have
been at this for a week now. How did you make it work in Asterisk 1.2x?



Rob








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RE: [asterisk-users] Alcatel OXO Sip

2006-09-19 Thread Christian Gatti

It the question why does asterisk has problems with SIP/2.0/udp or
SIP/2.0/UDP

if (strcasecmp(via, SIP/2.0/UDP)) {
ast_log(LOG_WARNING, Don't know how to respond via '%s'\n, via);
return -1;
}

This code says: I don't know what to do with a SIP/2.0/UDP in a via and
blocks (return -1).

Should the alcatel pbx not send something like SIP/2.0/udp in a via?

Christian


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Langstaff
Sent: Tuesday, September 19, 2006 13:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Alcatel OXO Sip

Oops! Yes, I was confusing strcmp and strcasecmp, sorry - too little coffee.

However, it's funny that the strcasecmp of SIP/2.0/UDP fails to match with
SIP/2.0/udp. Almost like it's using a strcmp instead of strcasecmp!

 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Christian Gatti

Asterisk gives me an WARNING if I try to register with my alcatel oxo
pbx.
Sep 18 11:27:19 WARNING[11977]: chan_sip.c:5259 check_via: Don't know 
how to respond via 'SIP/2.0/udp'

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Patrick
 Sent: 19 September 2006 10:57
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] Alcatel OXO Sip
 
 On Tue, 2006-09-19 at 02:18 -0700, Steve Langstaff wrote:
  I think that (one of the) offending line(s) is in chan_sip.c:
  
  if (strcasecmp(via, SIP/2.0/UDP)) {
  ast_log(LOG_WARNING, Don't know how to
 respond via '%s'\n, via);
  return -1;
  }
  
  This is looking for an upper-case 'UDP' whereas your oxo
 pbx is using
  a lower-case 'udp'.
 
 man strcasecmp says:
 strcasecmp, strncasecmp - compare two strings ignoring case
 
 So the case is ignored and either sip and SIP or udp and UDP should 
 work.
 
 Regards,
 Patrick
 
 
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[asterisk-users] polycom 501 digitmap

2006-09-19 Thread Jordan Novak




This is really 
starting to get to me. I have deleted this field in the phones per the wiki. I 
am trying to get the phones to dial on there own. Is there anyway to get the 
phone to dial 1-8 after three digits are received and 9 after seven to ten 
digits. I am willing to wait for a timeout but that doesn't seem to work. Any 
help is greatly appreciated.


Jordan Novak
Senior 
Telecommunications Engineer
Logistics Health Inc.


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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

Ryan wrote:
 Can you explain your design in a little more detail? What kind of hardware
 did you use to get over 1k users on a single box and 500 concurrent calls?
 Sounds like a very interesting medium-large scale implementation that
 others could learn from.
 
 thanks,
 Ryan 


I'll do the best I can from memory and without violating confidentiality :)

The build was for a startup ITSP and was the first of that scale that
either myself or my associate who worked for the client had done. The
hardware was something along these lines, but I cannot be absolutely sure:

3Ghz Dual XEON CPU
1GB RAM
2 1Gb NICs

I dont remember the hard drive specs at all, but that's more elementary
anyway.

We initially set up the systems with CentOS 4.2 or 4.3, can't remember.
MySQL 4.x (latest 4.x version from summer 2005)
Asterisk HEAD (constantly updating and recompiling, at the time the
realtime arch wasn't fully in place)
MySQL addons package
Realtime SIP clients
Statically configured SIP trunks, which provided our PSTN connections.
I cannot disclose the company, but the trunk provider is/was extremely
huge, a Tier 1 ISP.
MySQL CDRs (the cdr addon)
User options and feature controls accessed in realtime via a MySQL table
 designated for the purpose (basically an options table, with things
like call_forward (y/n) columns).
LOTS of custom monitoring done in regards to Asterisk status information
Custom PHP/MySQL/Apache web interface for provisioning, configuration,
and general administration written by yours truly, including polling
Asterisk for the status of a client UA when that client's config is
being viewed, provisioning (TFTP) handlers, etc...

Hope this is a good start, anything else you want to know, I'll do my best.

Also, once I finish my latest ITSP launch project, I'll be able to
(hopefully) give a better example, one with failover, custom CDRs,
custom LeastCost+BestPerformance routing, etc...etc... Even realtime
billing, which the previous client didn't have, AND reseller support at
the ITSP levelcan't say more yet, but it'll be rather huge I'm sure.
-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] Aastra 9133i and Atcom AT-320 - Comments please

2006-09-19 Thread Dave Cotton

On Tue, 2006-09-19 at 13:13 +0100, James Dyer wrote:
 I'm planning to deploy an Asterisk system in our office soon, and am 
 thinking of using a mixture of Aastra 9133i and Atcom AT-320 phones.
 
 Has anyone got any comments (good or bad) about these phone models?

I now only use Aastra phones, the 9133i is solid and professional
looking and works very well with *. My experience with support is A1.

Message waiting is well signalled as is no service.

The switch and POE save a lot of cabling. 


-- 
Dave Cotton [EMAIL PROTECTED]

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[asterisk-users] Problem with # locking up call

2006-09-19 Thread marvin horst

When I press # (once) during a bridged call it mutes/locks the call and I am unable to do anything on either end of the call.Then, if the called party hangs up before the calling party hangs up, the call still shows as connected with (show channels) even though both parties have hung up. Both channels are still locked and unable to initiate a new call until a soft hangup is performed.
If the calling party hangs up first nothing happens, but when the called party hangs up the channels are disconnected and free to make new calls.## still works to transfer the call, but not after the call gets locked.
Pressing # will not lock the call if I don't have the transfer options in the dial command.I'm using Asterisk SVN-branch-1.2-r42783 and#define ZAPTEL_VERSION 
SVN-branch-1.2-r1458*** show channel output Zap/9-1 [EMAIL PROTECTED]:1 Up Bridged Call(Zap/11-1)Zap/11-1 [EMAIL PROTECTED]:21 Up Dial(Zap/9|20|Tt) 
features.conf *[general]parkext = 700parkpos = 701-720context = parkedcallsparkingtime = 90featuredigittimeout = 500[featuremap]blindxfer = ##  ; Blind transfer
disconnect = *0 ; Disconnectautomon = *1   ; One Touch Recordatxfer = **   ; Attended transfer
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Re: [asterisk-users] How to install HUDLite Server

2006-09-19 Thread Zeeshan Zakaria

Thanks for your help. I'll send my future FOP questions to FOP mailing list. I am still having some troubles, like it doesn't show more than 34 extensions buttons, doesn't matter if button sizes are made smaller. Rest of the extensions don't show anywhere.
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Re: [asterisk-users] Problem with # locking up call

2006-09-19 Thread Michael Strelnikov

I had the same problem.The only solution I've found is to change blindxfer = ## to blindxfer = *#
 or something like that.I think there is a bug with that feature. It was working with some previous version but I don't remember which one.MichaelOn 9/19/06, 
marvin horst [EMAIL PROTECTED] wrote:
When I press # (once) during a bridged call it mutes/locks the call and I am unable to do anything on either end of the call.Then, if the called party hangs up before the calling party hangs up, the call still shows as connected with (show channels) even though both parties have hung up. Both channels are still locked and unable to initiate a new call until a soft hangup is performed.
If the calling party hangs up first nothing happens, but when the called party hangs up the channels are disconnected and free to make new calls.## still works to transfer the call, but not after the call gets locked.
Pressing # will not lock the call if I don't have the transfer options in the dial command.I'm using Asterisk SVN-branch-1.2-r42783 and#define ZAPTEL_VERSION 
SVN-branch-1.2-r1458*** show channel output Zap/9-1 [EMAIL PROTECTED]:1 Up Bridged Call(Zap/11-1)Zap/11-1 [EMAIL PROTECTED]:21 Up Dial(Zap/9|20|Tt) 
features.conf *[general]parkext = 700parkpos = 701-720context = parkedcallsparkingtime = 90featuredigittimeout = 500[featuremap]blindxfer = ##  ; Blind transfer
disconnect = *0 ; Disconnectautomon = *1   ; One Touch Recordatxfer = **   ; Attended transfer

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[asterisk-users] fast SIP failover (outgoing sIP requests) with 1.2

2006-09-19 Thread Klaus Darilion


Hi!

I have the following problem: I route calls from one office to the other 
office via SIP, but if for any reason this SIP call fails, I want a 
backup route via the PSTN.


Thus, I use:


exten =  _[1-9].,4,Dial(SIP/${enumresult},90)
exten =  _[1-9].,5,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?103:6)
exten =  _[1-9].,6,GotoIf($[${DIALSTATUS} = CONGESTION]?103:7)
exten =  _[1-9].,7,Hangup
exten =  _[1-9].,103,Dial(ZAP/g1/${EXTEN},90)

The problem is, if the SIP server at the remote office is down, thus no 
responses to the INVITE, it takes 64 seconds to timeout.


Is there a method to shorten this interval - e.g. if there is no 
response within 10 seconds give up - without changing the hardcoded 
retransmission value (6) in chan_sip ?


regards
klaus
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RE: [asterisk-users] fast SIP failover (outgoing sIP requests) wi th 1.2

2006-09-19 Thread Guido Hecken

 -Ursprüngliche Nachricht-
 Von: Klaus Darilion [mailto:[EMAIL PROTECTED]
 Gesendet: Dienstag, 19. September 2006 16:03
 An: asterisk-users@lists.digium.com
 Betreff: [asterisk-users] fast SIP failover (outgoing sIP requests) with
1.2
 
 Hi!
 
 I have the following problem: I route calls from one office to the other
 office via SIP, but if for any reason this SIP call fails, I want a
 backup route via the PSTN.
 
 Thus, I use:
 
 
 exten =  _[1-9].,4,Dial(SIP/${enumresult},90)
 exten =  _[1-9].,5,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?103:6)
 exten =  _[1-9].,6,GotoIf($[${DIALSTATUS} = CONGESTION]?103:7)
 exten =  _[1-9].,7,Hangup
 exten =  _[1-9].,103,Dial(ZAP/g1/${EXTEN},90)
 
 The problem is, if the SIP server at the remote office is down, thus no
 responses to the INVITE, it takes 64 seconds to timeout.
 
 Is there a method to shorten this interval - e.g. if there is no
 response within 10 seconds give up - without changing the hardcoded
 retransmission value (6) in chan_sip ?
 
 regards
 klaus

Hi,

maybe I'm wrong, but what about using the ChanisAvail function?

We did something like this in a customer installation:

exten = _XXX.,1,Set(LANGUAGE()=de)
exten = _XXX.,2,ChanisAvail(CAPI/ISDN3/${EXTEN},s)
exten = _XXX.,3,Dial(CAPI/ISDN3/${EXTEN}/b,60,tT)
exten = _XXX.,4,Congestion
exten = _XXX.,103,ChanisAvail(CAPI/ISDN2/${EXTEN},s)
exten = _XXX.,104,Dial(CAPI/ISDN2/${EXTEN}/b,60,tT)
exten = _XXX.,105,Congestion


Hope, it helps ...


Guido
 
gwsNetTech
Guido Hecken

Quirrenbacher Str. 36
53639 Königswinter
Germany

fon +49(2244) 870663
fax +49(2244) 870664
mobil  +49(179) 1267353
web http://www.gwsnettech.de
mailto:[EMAIL PROTECTED]

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Re: [asterisk-users] fast SIP failover (outgoing sIP requests) wi th 1.2

2006-09-19 Thread Klaus Darilion


Guido Hecken wrote:

-Ursprüngliche Nachricht-
Von: Klaus Darilion [mailto:[EMAIL PROTECTED]
Gesendet: Dienstag, 19. September 2006 16:03
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] fast SIP failover (outgoing sIP requests) with

1.2

Hi!

I have the following problem: I route calls from one office to the other
office via SIP, but if for any reason this SIP call fails, I want a
backup route via the PSTN.

Thus, I use:


exten =  _[1-9].,4,Dial(SIP/${enumresult},90)
exten =  _[1-9].,5,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?103:6)
exten =  _[1-9].,6,GotoIf($[${DIALSTATUS} = CONGESTION]?103:7)
exten =  _[1-9].,7,Hangup
exten =  _[1-9].,103,Dial(ZAP/g1/${EXTEN},90)

The problem is, if the SIP server at the remote office is down, thus no
responses to the INVITE, it takes 64 seconds to timeout.

Is there a method to shorten this interval - e.g. if there is no
response within 10 seconds give up - without changing the hardcoded
retransmission value (6) in chan_sip ?

regards
klaus


Hi,

maybe I'm wrong, but what about using the ChanisAvail function?

We did something like this in a customer installation:

exten = _XXX.,1,Set(LANGUAGE()=de)
exten = _XXX.,2,ChanisAvail(CAPI/ISDN3/${EXTEN},s)
exten = _XXX.,3,Dial(CAPI/ISDN3/${EXTEN}/b,60,tT)
exten = _XXX.,4,Congestion
exten = _XXX.,103,ChanisAvail(CAPI/ISDN2/${EXTEN},s)
exten = _XXX.,104,Dial(CAPI/ISDN2/${EXTEN}/b,60,tT)
exten = _XXX.,105,Congestion


Hope, it helps ...



Hi!

I've tried it but apparently chanisavail does not work with non-local 
SIP peers.


thanks
klaus
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Re: [asterisk-users] RE: FollowMe question

Hall, Eric M. wrote:
 
 I got the config working. Not sure if someone has pre-recorded sounds
 for this app or not. Looked all over for them and I'm unable to locate
 them.If anyone has sound file they would like to share that would help
 me greatly.
  
 Thanks
  
 
   *Sent:* Friday, September 15, 2006 5:23 PM
 *To:* 'asterisk-users@lists.digium.com'
 *Subject:* FollowMe question
 
 Group
  Does anyone have the FollowMe sound files? Do I need to record them?
 Also does anyone have a working followme.conf file that they would share?
 Thanks!
 
 
 
 
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I wouldn't mind a shot at creating the sound files in my little studio
here. Just give me a set of prompts/messages to record and I'll
contribute :)



-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] Polycom Expansion Module

2006-09-19 Thread Jerry Jones


Per 2.0.1 release notes

13315: Increased the maximum number of buddies to 8 for all platforms  
except

SoundPoint IP 600 and 601 which can watch 48 buddies



On Sep 18, 2006, at 10:35 PM, Douglas Garstang wrote:

48 was the limit on the number of speed dial entries that you could  
have in the directory. 7 was the old limit for the number of  
buddies you could watch. As far as I know, in 2.0.1, the number of  
entries you can have in the speed dial directory is 99, and the  
number of buddies that you can watch has gone up to 48.


Doug.

-Original Message-
From: Jerry Jones [mailto:[EMAIL PROTECTED]
Sent: Mon 9/18/2006 6:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [asterisk-users] Polycom Expansion Module



Poly 2.0.1 says it can do 48

On Sep 17, 2006, at 8:06 PM, Douglas Garstang wrote:

 As far as I know, it's 12.

   -Original Message-
   From: Noah Miller [mailto:[EMAIL PROTECTED]
   Sent: Sun 9/17/2006 10:27 AM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Cc:
   Subject: Re: [asterisk-users] Polycom Expansion Module



   Hi Kevin -

	Has anyone used the Polycom expansion module with  
multiple lines?

   
	My application is for 20 lines and read there was a  
limit of 7

 at one point.

	   I heard rumors that the newest version of the polycom sip  
firmware
	   (2.01) would lift the limit of 7.  It just came out, and I  
haven't

 had
   time to test it yet, but you can give it a try.

   - Noah
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Re: [asterisk-users] polycom 501 digitmap

2006-09-19 Thread Jerry Jones

the digitmap only tells the phone when to send the digits it has  
collected. They have no digit substitution feature. This would be  
done within your * dialplan



On Sep 19, 2006, at 7:57 AM, Jordan Novak wrote:

This is really starting to get to me. I have deleted this field in  
the phones per the wiki. I am trying to get the phones to dial on  
there own. Is there anyway to get the phone to dial 1-8 after three  
digits are received and 9 after seven to ten digits. I am willing  
to wait for a timeout but that doesn't seem to work. Any help is  
greatly appreciated.


Jordan Novak
Senior Telecommunications Engineer
Logistics Health Inc.
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[asterisk-users] transcoding error?

2006-09-19 Thread Damon Estep









Anyone encountered this on yet?



WARNING[23251]: chan_sip.c:2570 sip_write: Asked to transmit
frame type 64, while native formats is 4 (read/write = 4/4)



Started after an upgrade from CVS 8/2005 to current 1.2.12.1



If I had a reference for what frame types 4 and 64 are I might
be able to hunt this down, but I do not.



One suspicion is that it was on a MOHG.711u bridge
where the MOH format is quite MP3.



Any clues?






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Re: [asterisk-users] RE: FollowMe question

2006-09-19 Thread Darrick Hartman


Rushowr wrote:

Hall, Eric M. wrote:
  

I got the config working. Not sure if someone has pre-recorded sounds
for this app or not. Looked all over for them and I'm unable to locate
them.If anyone has sound file they would like to share that would help
me greatly.


snip--PLEASE learn to trim your replies


I wouldn't mind a shot at creating the sound files in my little studio
here. Just give me a set of prompts/messages to record and I'll
contribute :)
  
BJ already has sounds that are available for this app.  Since 1.4 has 
not been released, they may be a little harder to find than the sounds 
included with 1.2.  If you go here [1] you'll find sounds.  If they are 
not in that directory, I have the gsm files that BJ had originally 
included on the bug tracker when he originally created the app_followme 
for early 1.2 branch versions (note that it won't work in current 1.2 
versions but only SVN trunk (sometime to be 1.4)


[1]:  http://ftp.digium.com/pub/telephony/sounds/releases/

Darrick

--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
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Re: [asterisk-users] Aastra 9133i and Atcom AT-320 - Comments please

2006-09-19 Thread Jerry Jones

We tested a couple 9133i, dont remember the specifics right now but  
we stopped as there was some inconsistency in provisioning. I was  
very optimistic as I like the look and feel. We did deploy a couple  
480iCT which worked very well - when they worked. But they keep  
locking up and freezzing under heavy use, plus they have speakerphone  
and rfi issues.


You may wish to checkout the Polycom IP430, about the same price and  
have been very solid after installing 50 or so to date.



On Sep 19, 2006, at 8:15 AM, Dave Cotton wrote:


On Tue, 2006-09-19 at 13:13 +0100, James Dyer wrote:

I'm planning to deploy an Asterisk system in our office soon, and am
thinking of using a mixture of Aastra 9133i and Atcom AT-320 phones.

Has anyone got any comments (good or bad) about these phone models?


I now only use Aastra phones, the 9133i is solid and professional
looking and works very well with *. My experience with support is A1.

Message waiting is well signalled as is no service.

The switch and POE save a lot of cabling.


--
Dave Cotton [EMAIL PROTECTED]

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Re: [asterisk-users] fast SIP failover (outgoing sIP requests) wi th 1.2

2006-09-19 Thread Michiel van Baak

On 16:44, Tue 19 Sep 06, Klaus Darilion wrote:
 Hi!
 
 I've tried it but apparently chanisavail does not work with non-local 
 SIP peers.
 

In sip.conf try to add qualify=yes to the remote office part
This will make the sip peer dissapear when it's more then 2
seconds away (probably down). This is what we use for our
iax2 peers and it works great.
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?

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Re: [asterisk-users] transcoding error?


Damon Estep wrote:

Anyone encountered this on yet?

 


WARNING[23251]: chan_sip.c:2570 sip_write: Asked to transmit frame type
64, while native formats is 4 (read/write = 4/4)

 


Started after an upgrade from CVS 8/2005 to current 1.2.12.1

 


If I had a reference for what frame types 4 and 64 are I might be able
to hunt this down, but I do not.


show codecs in the Asterisk CLI.
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RE: [asterisk-users] Digium GUI?

2006-09-19 Thread Douglas Garstang

Could a carrier use it to provide service to it's customers, such that each 
customer only had access to their own content?

-Original Message-
From:   Steve Totaro [mailto:[EMAIL PROTECTED]
Sent:   Tue 9/19/2006 5:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: 
Subject:Re: [asterisk-users] Digium GUI?
Someone already said that they saw it at VON.  It was super simple to 
change the look and branding but the UI itself was nothing too special.

Thanks,
Steve

Douglas Garstang wrote:
 I wonder if the look and feel of this GUI will be completely configurable. If 
 it's not, then I really don't think that's very useful. Service providers 
 wouldn't be able to use it to let their customers manage their own settings, 
 and customers wouldn't want to use it if it wasn't branded with their company 
 info.

 \

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RE: [asterisk-users] fast SIP failover (outgoing sIP requests) with 1.2

2006-09-19 Thread Damon Estep

Try taking to 90 second timeout off

Change
exten =  _[1-9].,4,Dial(SIP/${enumresult},90)
to
exten =  _[1-9].,4,Dial(SIP/${enumresult})

a btter method is to set up each office as a unique peer with qualify = yes and 
then add the peer name to the dial string, like dial(SIP/[EMAIL PROTECTED])

if the peer is offline (qualify has failed) the unavaialbe status will come 
back right away.


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Klaus Darilion
 Sent: Tuesday, September 19, 2006 8:45 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] fast SIP failover (outgoing sIP requests)
 with 1.2
 
 Guido Hecken wrote:
  -Ursprüngliche Nachricht-
  Von: Klaus Darilion [mailto:[EMAIL PROTECTED]
  Gesendet: Dienstag, 19. September 2006 16:03
  An: asterisk-users@lists.digium.com
  Betreff: [asterisk-users] fast SIP failover (outgoing sIP requests)
 with
  1.2
  Hi!
 
  I have the following problem: I route calls from one office to the
 other
  office via SIP, but if for any reason this SIP call fails, I want a
  backup route via the PSTN.
 
  Thus, I use:
 
 
  exten =  _[1-9].,4,Dial(SIP/${enumresult},90)
  exten =  _[1-9].,5,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?103:6)
  exten =  _[1-9].,6,GotoIf($[${DIALSTATUS} = CONGESTION]?103:7)
  exten =  _[1-9].,7,Hangup
  exten =  _[1-9].,103,Dial(ZAP/g1/${EXTEN},90)
 
  The problem is, if the SIP server at the remote office is down, thus no
  responses to the INVITE, it takes 64 seconds to timeout.
 
  Is there a method to shorten this interval - e.g. if there is no
  response within 10 seconds give up - without changing the hardcoded
  retransmission value (6) in chan_sip ?
 
  regards
  klaus
 
  Hi,
 
  maybe I'm wrong, but what about using the ChanisAvail function?
 
  We did something like this in a customer installation:
 
  exten = _XXX.,1,Set(LANGUAGE()=de)
  exten = _XXX.,2,ChanisAvail(CAPI/ISDN3/${EXTEN},s)
  exten = _XXX.,3,Dial(CAPI/ISDN3/${EXTEN}/b,60,tT)
  exten = _XXX.,4,Congestion
  exten = _XXX.,103,ChanisAvail(CAPI/ISDN2/${EXTEN},s)
  exten = _XXX.,104,Dial(CAPI/ISDN2/${EXTEN}/b,60,tT)
  exten = _XXX.,105,Congestion
 
 
  Hope, it helps ...
 
 
 Hi!
 
 I've tried it but apparently chanisavail does not work with non-local
 SIP peers.
 
 thanks
 klaus
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[asterisk-users] Polycom default handset volume

2006-09-19 Thread Damon Estep









I had read a post somewhere that there is an XML parameter for
the Polycom config files for default handset volume, but I can not locate it
again.



Anyone know what it is?



I want to set the default handset volume higher on some
phones, despite the ADA hearing aid warning in the admin manual J






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Re: [asterisk-users] When does Scalability requests Asterisk

2006-09-19 Thread AK 4asterisk

SM  Sorry, should have been a little more specific. I've had Asterisk running realtime SIP users/peers andSM  realtime sql calls from the dialplan (all with MySQL), and have had around 2.5k registered users and a
SM  peak (that I recall) of around 500 concurrent calls.Wow that sounds pretty neat. Could you let us know what the HW specs were?- AK
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Re: [asterisk-users] fast SIP failover (outgoing sIP requests) wi th 1.2

2006-09-19 Thread Klaus Darilion


Michiel van Baak wrote:

On 16:44, Tue 19 Sep 06, Klaus Darilion wrote:

Hi!

I've tried it but apparently chanisavail does not work with non-local 
SIP peers.




In sip.conf try to add qualify=yes to the remote office part
This will make the sip peer dissapear when it's more then 2
seconds away (probably down). This is what we use for our
iax2 peers and it works great.


Ok. But what if there is no explicit peer entry in sip.conf? The fast 
failover should work not only when dialing the remote office (for which 
I have no peer entry as call routing is ENUM based) but for any SIP 
destination.


thanks
klaus
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[asterisk-users] Semi-OT: SIP or IAX provider in the Boston area?

2006-09-19 Thread Erik Anderson


Hello - I'll be heading out to the Boston area next week to start up a
branch office for my company.  I'll be implementing an Asterisk box as
part of their network infrastructure...so...does anyone have any
recommendations on a good reliable SIP or IAX provider?  I'd need DIDs
for incoming calls as well.

Thanks!
-Erik

--
Erik Anderson
http://andersonfam.org
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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

[EMAIL PROTECTED] wrote:

 Can you explain your design in a little more detail? What kind of hardware
 did you use to get over 1k users on a single box and 500 concurrent calls?
 Sounds like a very interesting medium-large scale implementation that
 others could learn from.
 
 thanks,
 Ryan

(NOTE: I sent the original reply about 3 hours ago and have not seen it
post, so I'm resending. I apologize for any double receipts of the message.)

I'll do the best I can from memory and without violating confidentiality
 :)

The build was for a startup ITSP and was the first of that scale that
either myself or my associate who worked for the client had done. The
hardware was something along these lines, but I cannot be absolutely sure:

3Ghz Dual XEON CPU
1GB RAM
2 1Gb NICs

I dont remember the hard drive specs at all, but that's more elementary
anyway.

We initially set up the systems with CentOS 4.2 or 4.3, can't remember.
MySQL 4.x (latest 4.x version from summer 2005)
Asterisk HEAD (constantly updating and recompiling, at the time the
realtime arch wasn't fully in place)
MySQL addons package
Realtime SIP clients
Statically configured SIP trunks, which provided our PSTN connections.
I cannot disclose the company, but the trunk provider is/was extremely
huge, a Tier 1 ISP.
MySQL CDRs (the cdr addon)
User options and feature controls accessed in realtime via a MySQL table
 designated for the purpose (basically an options table, with things
like call_forward (y/n) columns).
LOTS of custom monitoring done in regards to Asterisk status information
Custom PHP/MySQL/Apache web interface for provisioning, configuration,
and general administration written by yours truly, including polling
Asterisk for the status of a client UA when that client's config is
being viewed, provisioning (TFTP) handlers, etc...

Hope this is a good start, anything else you want to know, I'll do my best.

Also, once I finish my latest ITSP launch project, I'll be able to
(hopefully) give a better example, one with failover, custom CDRs,
custom LeastCost+BestPerformance routing, etc...etc... Even realtime
billing, which the previous client didn't have, AND reseller support at
the ITSP levelcan't say more yet, but it'll be rather huge I'm sure.


-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] When does Scalability requests Asterisk

2006-09-19 Thread AK 4asterisk

Never mind my previous question. I see that Ryan already asked that and you responded. It does sound like a pretty neat project. Would love to hear more once you finish the project that you are working on currently.
- AK
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[asterisk-users] Call forward with CFU?

2006-09-19 Thread Roy Sigurd Karlsbakk


Hi all

I've got the following message from the telco regarding call forward  
number presentation. Can someone please help me decipher this? I  
don't understand shit about this :P


roy



Roy ,

Below is an extract taken from a working scenario of the CFU (A B -  
B C) functionality problem :


Please can you activate the Asterisks equilivant  = Facility- 
Information-Element-Component.


Let us know how u get on



  --- 1. Q932 FAC IE ---
Invoke ID  : 0011Operation: 000FComponent : 00A1
Data offset: 0008CallRef  : 0005CRFlag: 
Whole data: A1 21 02 01 11 02 01 0F 30 19 02 01 01 0A 01 01
A1 11 A0 0F A1 0D 0A 01 02 12 08 32 31 39 33 32
31 31 30
-
Facility-Information-Element-Components Components ::=
  invokeComp:
invokeID17
operation-value localValue: 
divertingLegInformation2  {c

SEQUENCE
  INTEGER   1
  ENUMERATED   1
  [1]
[0]
  [1]
ENUMERATED   2
NumericString   21932110




---
Humans mostly aren't particularly evil. They just get carried away  
by new ideas, like dressing up in jackboots and shooting people, or  
dressing up in white sheets and lynching people, or dressing up in  
tie-dye jeans and playing guitars at people

 - Terry Pratchett
---
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]



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Re: [asterisk-users] Problem with # locking up call

2006-09-19 Thread Anthony Cennami

I've also found similar problems with blindxfer -- such as when someone is attempting to interact with an IVR using a '#' option. By default # seems to transfer a call, but if you have blindxfer enabled with '#1' or ##, then Asterisk hears the first # and waits for a following DTMF.
At the moment I have the option disabled in feature map, since # works by default anyways.On 9/19/06, Michael Strelnikov 
[EMAIL PROTECTED] wrote:I had the same problem.The only solution I've found is to change 
blindxfer = ## to blindxfer = *#
 or something like that.I think there is a bug with that feature. It was working with some previous version but I don't remember which one.Michael
On 9/19/06, 
marvin horst [EMAIL PROTECTED] wrote:

When I press # (once) during a bridged call it mutes/locks the call and I am unable to do anything on either end of the call.Then, if the called party hangs up before the calling party hangs up, the call still shows as connected with (show channels) even though both parties have hung up. Both channels are still locked and unable to initiate a new call until a soft hangup is performed.
If the calling party hangs up first nothing happens, but when the called party hangs up the channels are disconnected and free to make new calls.## still works to transfer the call, but not after the call gets locked.
Pressing # will not lock the call if I don't have the transfer options in the dial command.I'm using Asterisk SVN-branch-1.2-r42783 and#define ZAPTEL_VERSION 
SVN-branch-1.2-r1458*** show channel output Zap/9-1 [EMAIL PROTECTED]:1 Up Bridged Call(Zap/11-1)Zap/11-1 [EMAIL PROTECTED]:21 Up Dial(Zap/9|20|Tt) 
features.conf *[general]parkext = 700parkpos = 701-720context = parkedcallsparkingtime = 90featuredigittimeout = 500[featuremap]blindxfer = ##  ; Blind transfer
disconnect = *0 ; Disconnectautomon = *1   ; One Touch Recordatxfer = **   ; Attended transfer

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http://lists.digium.com/mailman/listinfo/asterisk-users-- Anthony D Cennami
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RE: [asterisk-users] Polycom default handset volume

2006-09-19 Thread Douglas Garstang

I don't think you can set a default volume, but you can configure the handset 
(and headset) volume to persist between calls. Look for 'persist' in sip.cfg or 
phone1.cfg.

Doug.


-Original Message-
From:   Damon Estep [mailto:[EMAIL PROTECTED]
Sent:   Tue 9/19/2006 9:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: 
Subject:[asterisk-users] Polycom default handset volume
I had read a post somewhere that there is an XML parameter for the
Polycom config files for default handset volume, but I can not locate it
again.

 

Anyone know what it is?

 

I want to set the default handset volume higher on some phones, despite
the ADA hearing aid warning in the admin manual :-)




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RE: [asterisk-users] Digium GUI?

2006-09-19 Thread shadowym


I am talking about the GUI that was announced as part of the new Asterisk
Appliance.

Sounds like it is going to be a full featured GUI like FreePBX.
 

-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED] 
Sent: Monday, September 18, 2006 8:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Digium GUI?

 So the press announcement said that the new Digium GUI will be 
 available in
 v1.4 sometime in Oct.  Is the GUI already there in Trunk or is there 
 some other branch of development that the general public cannot access?

Do you mean this?

http://svn.digium.com/view/asterisk/trunk/static-http/



On 9/18/06, Don Fanning [EMAIL PROTECTED] wrote:
 You mean the menuselect ncurses screen?  If yes, then yes... it's a 
 gui. :)

 -Original Message-
 From: shadowym [mailto:[EMAIL PROTECTED]
 Sent: Monday, September 18, 2006 4:43 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Digium GUI?


 So the press announcement said that the new Digium GUI will be 
 available in
 v1.4 sometime in Oct.  Is the GUI already there in Trunk or is there 
 some other branch of development that the general public cannot access?

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Re: [asterisk-users] Polycom default handset volume

2006-09-19 Thread Jessee J Holmes

I think the following should resolve your problem, if I'm understanding it correctly:In the sip.cfg file, look at setting the following:volume voice.volume.persist.handset="1" voice.volume.persist.headset="1" voice.volume.persist.handsfree="1"/1= remember last setting, 0=return to defaultThis will make the phone remember your setting and will not reset the setting every time you need to make a new call. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 19, 2006, at 10:23 AM, Damon Estep wrote: I had read a post somewhere that there is an XML parameter for the Polycom config files for default handset volume, but I can not locate it again. Anyone know what it is? I want to set the default handset volume higher on some phones, despite the ADA hearing aid warning in the admin manual J ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [asterisk-users] Problem with # locking up call


Anthony Cennami wrote:

I've also found similar problems with blindxfer -- such as when someone is
attempting to interact with an IVR using a '#' option.  By default # seems
to transfer a call, but if you have blindxfer enabled with '#1' or ##, then
Asterisk hears the first # and waits for a following DTMF.


Only if you use t or T option to Dial when calling IVRs
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Re: [asterisk-users] Alcatel OXO Sip

2006-09-19 Thread Brian Candler

On Tue, Sep 19, 2006 at 02:53:17PM +0200, Christian Gatti wrote:
 It the question why does asterisk has problems with SIP/2.0/udp or
 SIP/2.0/UDP
 
 if (strcasecmp(via, SIP/2.0/UDP)) {
   ast_log(LOG_WARNING, Don't know how to respond via '%s'\n, via);
   return -1;
 }
 
 This code says: I don't know what to do with a SIP/2.0/UDP in a via and
 blocks (return -1).

No, it says I don't know what to do if the via is *not* SIP/2.0/UDP
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Re: [Asterisk-Users] How to learn or teach VoIP QoE

2006-09-19 Thread Mojo with Horan Company, LLC

I've heard a MOS of 4.4 is ulaw/alaw, presumably like a local exchange 
call through the pstn.


testyourvoip.com tells me that the highest score available with G.729 is 
4.2, which is pretty darn close to 4.4.


I don't know why I think this (or why I've heard it (or if it's right)) 
but I think gsm is 3.8?  Maybe someone can confirm or disconfirm this. 
This kinda seems like the codec my long distance calls go out on but I'm 
not really sure.


http://www.testyourvoip.com/results.html?id=071GM0result=0
(one of my more POOR results) approximates about 2.75 for 'tin cans and 
string', 3.2 for a crummy cell phone call, about 3.9 for a decent cell 
call, 4.4 or so for 'like calling next door', and the mystical 5.0 for 
'better than being there'.


I wonder if 16kHz wideband codecs would bring our voice-carrying 
experiences into the 5.0 range?


Moj



Olivier wrote:

Hi,

How would you best learn VoIP Quality of Experience ?

Before diving into packet loss and jitter, I would like to know what a 
toll-quality call is, what a rated 3.5 MOS call is like.

I'm wondering how I should proceed.

Shall I :
- get pre-recorded sound files somewhere and simply stream them to a MOS 
enabled softphone (Counterpath sells eye-beam which includes a telchemy 
MOS rating module),
- or shall I install some network impairment software, generate VoIP 
trafic and tweak myself jitter and other parameters so that I can 
associate network measures to call quality ?


I've never heard of any sound files library aimed to learn what the 
impact of packet is like for end user experience.
I've seen here and there network simulators (some of them free of 
charge) but it seems tricky to tune them to VoIP (is a 10% packet loss 
realistic or not ?).


To make myself perfectly clear, my ultimate goal is to better undestand 
users testimonies when they warn me about poor quality phone calls.


Regards
!DSPAM:500,450f2da465292693510148!




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!DSPAM:500,450f2da465292693510148!


--
Mojo [EMAIL PROTECTED]
Office Manager, Horan  Company, LLC
(907) 747- x112
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[asterisk-users] mpg123

2006-09-19 Thread Giordano Grandis




Hi 
all,
I'm using * 1.0.9 
which use mpg123 for music on hold. But sometimes starts eating up a lot of 
CPU.
I sthere any 
alternative method to use moh without use mpg123?
I tryied this http://astrecipes.net/?n=152but i 
doesn't wotks for me.

Anyone can help me 
pls ?

Thanks in 
advance.

Giordano
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Re: [asterisk-users] Problem with # locking up call

2006-09-19 Thread Anthony Cennami

Which we use on all calls so that if someone wants to transfer said in/out call they can (such as a secretary placing a call for an executive, or a regular user transfering a call to a verification service/person.)
On 9/19/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Anthony Cennami wrote: I've also found similar problems with blindxfer -- such as when someone is attempting to interact with an IVR using a '#' option.By default # seems to transfer a call, but if you have blindxfer enabled with '#1' or ##, then
 Asterisk hears the first # and waits for a following DTMF.Only if you use t or T option to Dial when calling IVRs___--Bandwidth and Colocation provided by 
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-- Anthony D Cennami
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Re: [asterisk-users] When does Scalability requests Asterisk

2006-09-19 Thread Roy Sigurd Karlsbakk

SM  Sorry, should have been a little more specific. I've had  
Asterisk running realtime SIP users/peers and
SM  realtime sql calls from the dialplan (all with MySQL), and  
have had around 2.5k registered users and a

SM  peak (that I recall) of around 500 concurrent calls.

Wow that sounds pretty neat. Could you let us know what the HW  
specs were?


The tests we've done shows that asterisk doing RTP bridging SIP/SIP  
calls can handle up to approxmately 4-500 calls for a single Xeon 3.0  
before locking up, spending approx 60-70% system/kernel time, _not_  
usertime. We have not measured when audio quality starts to suffer,  
but I would guess that happens around 300 or so. If you're allowed to  
use reinvites (not having clients behind NAT and so on), the number  
obviously climbes.


Note: NO you can NOT use reinvites for clients behind NAT in my  
scenario: Several trunks/pstn-gateways talking SIP to a hub server  
talking to clients. Clients register with hub server. pstngw gets a  
call in, sends it to hub server, hub server sends reinvite to pstngw,  
pstngw sends invite to client whose NAT gateway does not know the  
pstngw's address and throws the packet away...


roy
---
Humans mostly aren't particularly evil. They just get carried away  
by new ideas, like dressing up in jackboots and shooting people, or  
dressing up in white sheets and lynching people, or dressing up in  
tie-dye jeans and playing guitars at people

 - Terry Pratchett
---
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]



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[asterisk-users] codecs/voicemail/DTMF

2006-09-19 Thread Mr. Jones


Hi Folks,

We're trying to roll Asterisk out to production and are having a few
complications.

Most specifically we have G711 for our inbound origination, but would
prefer G729 for outbound termination, so far so good - it appears that
dtmfmode=auto works in both cases.

The area I'm having trouble with is, in order to have g729 on the
outbound I have:

disallow=all
allow=g729
allow=ulaw
allow=alaw

In sip.conf at the [general] level.

When we call voicemail, or the auto attendant internally touchtones
don't work and we get:

WARNING[8393]: dsp.c:1422 ast_dsp_process: Inband DTMF is not
supported on codec g729. Use RFC2833

I'm just guessing, but I thought auto was supposed to negotiate the
DTMF mode. Since it appears that the voicemail can't handle RFC2833,
is there some way to force the codec to resort to G711?

Thanks!

Brian
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[asterisk-users] SIP Lines Example Citel

2006-09-19 Thread Steve Totaro

Anyone know how to setup the SIP lines on a Citel box so it can register 
with Asterisk.  I keep getting Unauthorized and I have tried every 
different combination of settings that I can think of.  I am not sure 
what fields are required or what information goes where in the Citel 
interface.


Thanks,
Steve Totaro
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[asterisk-users] 404 not Found

2006-09-19 Thread harrygaillac-sip

Hello,


I wish asterisk to forward the none local uri to an
outbound proxy instead of send back 404 .

I add this in extension.conf and outbound is a peer
with an outbounproxy set to SER :

[sip]

exten = _.,1,NoOp(Incoming Call from house extension
${CALLERID} for [EMAIL PROTECTED])
exten = _.,2,GotoIf($[${SIPDOMAIN} = nxs.yi.org]?3:4)
exten = _.,3,Goto(sip-local,${EXTEN},1)
exten = _.,4 Goto(outbound)
exten = h,1,HangUp()

[outbound]
exten = _.,1,Dial(Sip/outbound)

[sip-local]



Harry








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Re: [asterisk-users] When does Scalability requests Asterisk

2006-09-19 Thread Jay R. Ashworth

On Tue, Sep 19, 2006 at 07:01:41PM +0200, Roy Sigurd Karlsbakk wrote:
 The tests we've done shows that asterisk doing RTP bridging SIP/SIP  
 calls can handle up to approxmately 4-500 calls for a single Xeon 3.0  
 before locking up, spending approx 60-70% system/kernel time, _not_  
 usertime. We have not measured when audio quality starts to suffer,  
 but I would guess that happens around 300 or so. If you're allowed to  
 use reinvites (not having clients behind NAT and so on), the number  
 obviously climbes.

Newbie question: that's if all the audio is passing over the server's
bus, right?

I'm looking at a pretty big system using either SIP or MGCP to tell a
bunch of FXS and T-1 media gateway boxes to talk to each other over a
dedicated GigE -- would that *be* a reinvite situation, generally, or
not?  I'm assuming that since the server would only be doing MoH, VRX,
and the like, that I'm in much better shape loadwise, even at 40xSIP +
390xFXS.

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth  AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
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[asterisk-users] Re: Semi-OT: SIP or IAX provider in the Boston area?

2006-09-19 Thread Erik Anderson


On 9/19/06, Erik Anderson [EMAIL PROTECTED] wrote:

Hello - I'll be heading out to the Boston area next week to start up a
branch office for my company.  I'll be implementing an Asterisk box as
part of their network infrastructure...so...does anyone have any
recommendations on a good reliable SIP or IAX provider?  I'd need DIDs
for incoming calls as well.


My apologies for posting this business-related query to the
non-commercial list.  It won't happen again :-)

-Erik
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[asterisk-users] Repost: Register message received from realtime peer crashes Asterisk

2006-09-19 Thread Cameron and Karlene

When Asterisk (1.2.12.1) receives a SIP register message for a realtimepeer, the CLI reports "Disconnected from Asterisk server". Asterisk hasdisappeared:asterisk -rUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctlexist?)A look at the full log doesn't reveal much:Sep 17 06:11:25 DEBUG[11011] acl.c: # Testing 60.234.nnn.nnn with192.168.1.0Sep 17 06:11:25 DEBUG[11011] chan_sip.c: Target address 60.234.nnn.nnn isnot local, substituting externipSep 17 06:11:25 DEBUG[11011] res_config_mysql.c: MySQL RealTime: RetrieveSQL: SELECT * FROM sip_buddies WHERE name = '6000'Sep 17 06:11:25 DEBUG[11011] res_config_mysql.c: MySQL RealTime: Everythingis fine.Asterisk then restarts (it gets a new pid) and will continue running
 happilyuntil a new register request is received for a realtime peer. Note thatAsterisk operates normally in all other respects until the register isreceived e.g. sip peers in sip.conf can register and make callssuccessfully. Only when a register is received from a peer that exists insip_buddies does Asterisk crash.I can run the query successfully on mysql command line:SELECT * FROM sip_buddies WHERE name = '6000';snip1 row in set (0.32 sec)A review of syslog and the mysql log reveals little:mysql log060917 6:54:31 14 Init DB asterisk 14 Query SELECT * FROM sip_buddies WHERE name = '6000'syslogNothing report at the time of the crash (06:54).extconfig.conf[settings]sipusers = mysql,asterisk,sip_buddiessippeers =
 mysql,asterisk,sip_buddiesres_mysql.conf[general]dbhost = 127.0.0.1dbname = asteriskdbuser = rootdbpass = passworddbport = 3306Could anyone advise what's going on or further checking that I could do toanalyse the problem?ThanksCameron
		 
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Re: [asterisk-users] codecs/voicemail/DTMF

Use type=user for inbound and type=peer for outbound.  Have different 
codec settings for each of them.


Mr. Jones wrote:

Hi Folks,

We're trying to roll Asterisk out to production and are having a few
complications.

Most specifically we have G711 for our inbound origination, but would
prefer G729 for outbound termination, so far so good - it appears that
dtmfmode=auto works in both cases.

The area I'm having trouble with is, in order to have g729 on the
outbound I have:

disallow=all
allow=g729
allow=ulaw
allow=alaw

In sip.conf at the [general] level.

When we call voicemail, or the auto attendant internally touchtones
don't work and we get:

WARNING[8393]: dsp.c:1422 ast_dsp_process: Inband DTMF is not
supported on codec g729. Use RFC2833

I'm just guessing, but I thought auto was supposed to negotiate the
DTMF mode. Since it appears that the voicemail can't handle RFC2833,
is there some way to force the codec to resort to G711?

Thanks!

Brian
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Re: [Asterisk-Users] How to learn or teach VoIP QoE

2006-09-19 Thread Steve Underwood


Mojo with Horan  Company, LLC wrote:

I've heard a MOS of 4.4 is ulaw/alaw, presumably like a local exchange 
call through the pstn.


testyourvoip.com tells me that the highest score available with G.729 
is 4.2, which is pretty darn close to 4.4.


Alaw and ulaw are about 4.4. ulaw on a robbed bit trunk is about 4.3. 
G.729 is about 3.9. That doesn't sound like a big spread, but MOS tends 
to bunch them together. G.729 at 3.9 is pretty reasonable, while 
anything scoring 3.0 is bloody awful.




I don't know why I think this (or why I've heard it (or if it's 
right)) but I think gsm is 3.8?  Maybe someone can confirm or 
disconfirm this. This kinda seems like the codec my long distance 
calls go out on but I'm not really sure.


3.8 sounds about right.



http://www.testyourvoip.com/results.html?id=071GM0result=0
(one of my more POOR results) approximates about 2.75 for 'tin cans 
and string', 3.2 for a crummy cell phone call, about 3.9 for a decent 
cell call, 4.4 or so for 'like calling next door', and the mystical 
5.0 for 'better than being there'.


I wonder if 16kHz wideband codecs would bring our voice-carrying 
experiences into the 5.0 range?


Much of the reason ulaw/alaw is way below 5.0 is due to bandwidth 
limitations. 16kHz sampling gets much closer to 5.0 It takes more like 
32kHz sampling to actually reach 5.0, though.


Steve


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[asterisk-users] gTalk no audio issue

2006-09-19 Thread Alex Guan

Gang,With the latest code I am having audio issue with the chan_gtalk and wondering if anyone else had the same problem- I was able to set up the call between Asterisk and my gTalk account, but there was no audio
- Looking closer, I am seeing these messages for an incoming call: -- SIP/5001-081ef020 is ringing[Sep 19 12:13:44] DEBUG[31226]: rtp.c:1402 ast_rtp_early_bridge: Channel 'Gtalk/guan.alex-e086' 
has no RTP, not doing anything[Sep 19 12:13:44] NOTICE[31226]: chan_gtalk.c:1331 gtalk_indicate: Don't know how to indicate condition '3'[Sep 19 12:13:44] DEBUG[31226]: channel.c
:2278 ast_indicate_data: Driver for channel 'Gtalk/guan.alex-e086' does not support indication 3, emulating it[Sep 19 12:13:44] DEBUG[31226]: channel.c:2427 ast_prod: Prodding channel 'Gtalk/guan.alex-e086'[Sep 19 12:13:44] DEBUG[31226]: 
channel.c:2652 set_format: Set channel Gtalk/guan.alex-e086 to write format slinJABBER: asterisk INCOMING: iq to=
[EMAIL PROTECTED]/asterisk9642378C type=set id=98 from=[EMAIL PROTECTED]/Talk.v96A3F055BCsession type=transport-info id=1314397402 initiator=
[EMAIL PROTECTED]/Talk.v96A3F055BC xmlns=http://www.google.com/sessiontransport xmlns=
http://www.google.com/transport/p2pcandidate name=rtp address=10.10.150.96 port=4923 preference=1 username=NXkxfCIYx2p8tMNc protocol=udp generation=0 password=mVSwEuvfiU9y062J type=local network=0//transport/session/iq
 -- JABBER: I Dont have an IQ!!!Does anybody know why there is no RTP? What am I missing here?And here is my gtalk.conf
:[general]context=gtalkallowguest=yes [guest] disallow=allallow=ulawcontext=guest[guan.alex]username=
[EMAIL PROTECTED] disallow=allallow=ulawallow=ilbcallow=isaccontext=gtalkconnection=asterisk My jabber.conf:
[general]debug=yes autoprune=yes autoregister=yes [asterisk] 
type=client serverhost=talk.google.com username=[EMAIL PROTECTED] ;;secret=xx  
port=5222 usetls=yes ;usesasl=yes buddy=[EMAIL PROTECTED] statusmessage=online 
timeout=100 Your help is greatly appreciated!Thanks,Alex
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[asterisk-users] Re: gTalk no audio issue

2006-09-19 Thread Alex Guan

I should have posted the logs when the call is acceptedhere it is:-- SIP/5001-081ef020 answered Gtalk/guan.alex-e086[Sep 19 12:13:47] DEBUG[31226]: rtp.c:1402 ast_rtp_early_bridge: Channel 'Gtalk/guan.alex-e086' 
has no RTP, not doing anything[Sep 19 12:13:47] NOTICE[31226]: chan_gtalk.c:1331 gtalk_indicate: Don't know how to indicate condition '-1'[Sep 19 12:13:47] DEBUG[31226]: channel.c
:2652 set_format: Set channel Gtalk/guan.alex-e086 to write format ulaw[Sep 19 12:13:47] DEBUG[31226]: chan_gtalk.c:513 gtalk_answer: Answer!JABBER: asterisk OUTGOING: iq type='set' to='
[EMAIL PROTECTED]/Talk.v96A3F055BC' from='[EMAIL PROTECTED]/asterisk9642378C' id='i'session xmlns='http://www.google.com/session
' type='accept' initiator='[EMAIL PROTECTED]/Talk.v96A3F055BC' id='1314397402'description xmlns='http://www.google.com/session/phone
' xml:lang='en'payload-type id='0' name='PCMU' clockrate='8000' bitrate='64000'/payload-type id='100' name='EG711U' clockrate='8000' bitrate='64000'/payload-type id='102' name='iLBC' clockrate='8000' bitrate='13300'/payload-type id='106' name='telephone-event' clockrate='8000'//descriptiontransport xmlns='
http://www.google.com/transport/p2p'//session/iq[Sep 19 12:13:47] WARNING[31226]: rtp.c:3019 ast_rtp_bridge: Can't find native functions for channel 'Gtalk/guan.alex-e086'
 -- Native bridging Gtalk/guan.alex-e086 and SIP/5001-081ef020 endedThanks again,AlexOn 9/19/06, Alex Guan 
[EMAIL PROTECTED] wrote:Gang,With the latest code I am having audio issue with the chan_gtalk and wondering if anyone else had the same problem
- I was able to set up the call between Asterisk and my gTalk account, but there was no audio
- Looking closer, I am seeing these messages for an incoming call: -- SIP/5001-081ef020 is ringing[Sep 19 12:13:44] DEBUG[31226]: rtp.c:1402 ast_rtp_early_bridge: Channel 'Gtalk/guan.alex-e086' 
has no RTP, not doing anything[Sep 19 12:13:44] NOTICE[31226]: chan_gtalk.c:1331 gtalk_indicate: Don't know how to indicate condition '3'[Sep 19 12:13:44] DEBUG[31226]: channel.c

:2278 ast_indicate_data: Driver for channel 'Gtalk/guan.alex-e086' does not support indication 3, emulating it[Sep 19 12:13:44] DEBUG[31226]: channel.c:2427 ast_prod: Prodding channel 'Gtalk/guan.alex-e086'[Sep 19 12:13:44] DEBUG[31226]: 
channel.c:2652 set_format: Set channel Gtalk/guan.alex-e086 to write format slinJABBER: asterisk INCOMING: iq to=

[EMAIL PROTECTED]/asterisk9642378C type=set id=98 from=
[EMAIL PROTECTED]/Talk.v96A3F055BCsession type=transport-info id=1314397402 initiator=
[EMAIL PROTECTED]/Talk.v96A3F055BC xmlns=
http://www.google.com/sessiontransport xmlns=
http://www.google.com/transport/p2pcandidate name=rtp address=
10.10.150.96 port=4923 preference=1 username=NXkxfCIYx2p8tMNc protocol=udp generation=0 password=mVSwEuvfiU9y062J type=local network=0//transport/session/iq
 -- JABBER: I Dont have an IQ!!!Does anybody know why there is no RTP? What am I missing here?And here is my gtalk.conf

:[general]context=gtalkallowguest=yes [guest] disallow=allallow=ulawcontext=guest[guan.alex]username=
[EMAIL PROTECTED] disallow=allallow=ulawallow=ilbcallow=isaccontext=gtalkconnection=asterisk 
My jabber.conf:
[general]debug=yes autoprune=yes autoregister=yes [asterisk] 
type=client serverhost=talk.google.com username=
[EMAIL PROTECTED] ;;secret=xx  
port=5222 usetls=yes ;usesasl=yes buddy=
[EMAIL PROTECTED] statusmessage=online 
timeout=100 Your help is greatly appreciated!Thanks,Alex


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Re: [asterisk-users] Problem with # locking up call