date:20130321

[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

2013-03-21 Thread Florian Wolters

Hi @ll,

I just moved my Asterisk Box and changed the Provider and Internet Access to a 
full IP Access by Deutsche Telekom.

I set up my sip.conf as I found various examples throughout the Net. Calls and 
some other stuff is basically working. 

The problem I ran into is, that the outgoing and incoming calls are dropped 
after exactly 15 Minutes. Solution for this should be setting the 
session-timers to refuse but this doesnt change anything here. 

I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest 
Asterisk by Digium without success. 

Has anyone else has the Same problem or is a solution already known? Could 
someone point me in the right direction? I can provide (debug) logs if 
essential.

Best regards

   Flo


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Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

2013-03-21 Thread isrlgb

Try canreinvite=yes in sip trunk

-Original Message-
From: Florian Wolters 
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 21 Mar 2013 08:31:54 
To: 
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

Hi @ll,

I just moved my Asterisk Box and changed the Provider and Internet Access to a 
full IP Access by Deutsche Telekom.

I set up my sip.conf as I found various examples throughout the Net. Calls and 
some other stuff is basically working. 

The problem I ran into is, that the outgoing and incoming calls are dropped 
after exactly 15 Minutes. Solution for this should be setting the 
session-timers to refuse but this doesnt change anything here. 

I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest 
Asterisk by Digium without success. 

Has anyone else has the Same problem or is a solution already known? Could 
someone point me in the right direction? I can provide (debug) logs if 
essential.

Best regards

   Flo

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Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

2013-03-21 Thread Leandro Dardini

2013/3/21 Florian Wolters :
> Hi @ll,
>
> I just moved my Asterisk Box and changed the Provider and Internet Access to 
> a full IP Access by Deutsche Telekom.
>
> I set up my sip.conf as I found various examples throughout the Net. Calls 
> and some other stuff is basically working.
>
> The problem I ran into is, that the outgoing and incoming calls are dropped 
> after exactly 15 Minutes. Solution for this should be setting the 
> session-timers to refuse but this doesnt change anything here.
>
> I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest 
> Asterisk by Digium without success.
>
> Has anyone else has the Same problem or is a solution already known? Could 
> someone point me in the right direction? I can provide (debug) logs if 
> essential.
>
> Best regards
>
>Flo
>
>

I think it is important to know the reason the call is disconnected.
Start checking who is sending the BYE and if before the BYE there is
other weird packets, like retry of packet sending ...

A simple "tcpdump" can help explain all the mistery.

Leandro

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Re: [asterisk-users] Laptop error

2013-03-21 Thread Frederic Van Espen

On Mon, 2013-03-11 at 14:34 +0100, Patrick Lists wrote:
> On 03/11/2013 12:53 PM, termo termosel wrote:
> > Hi,
> >
> > I have Ubuntu and Asterisk 11.2.1 in a boot USB. When I put it in
> > desktop computer, asterisk starts without problem but if I insert
> the
> > same USB in a laptop computer Asterisk doesn't start. Is it possible
> > because different microprocessors?
> 
> Yes. If you made the USB stick on a x86_64 (64 bit) computer and then 
> try it on a x86 (32 bit) laptop, it will not work. 

Still weird. If the OS is x86_64, then asterisk should have no problem
starting up if it was compiled for x86_64. If you try to boot the x86_64
os on a x86 only CPU, the OS would not boot at all.

Perhaps asterisk was compiled with some extra instruction flags that are
only supported on the desktop computer CPU though.

Frederic


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Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

2013-03-21 Thread Zyumbilev, Peter

I had this exact problem with my voip provider a few years ago.

It was disconnecting at exactly 5 minutes.

I solved it by moving Asterisk 1.6 to Asterisk 1.4.

Try asterisk 1.4 or 1.8  on a test box and see how it goes.

Peter

On 21/03/2013 09:31, Florian Wolters wrote:
> Hi @ll,
> 
> I just moved my Asterisk Box and changed the Provider and Internet Access to 
> a full IP Access by Deutsche Telekom.
> 
> I set up my sip.conf as I found various examples throughout the Net. Calls 
> and some other stuff is basically working. 
> 
> The problem I ran into is, that the outgoing and incoming calls are dropped 
> after exactly 15 Minutes. Solution for this should be setting the 
> session-timers to refuse but this doesnt change anything here. 
> 
> I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest 
> Asterisk by Digium without success. 
> 
> Has anyone else has the Same problem or is a solution already known? Could 
> someone point me in the right direction? I can provide (debug) logs if 
> essential.
> 
> Best regards
> 
>Flo
> 
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> 
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> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

2013-03-21 Thread Robert Krakora

I am having the same problem with Asterisk 11.2.0 and Linphone and it is
exactly 15 minutes and occurring with SIP running on our LAN.

On Thu, Mar 21, 2013 at 3:31 AM, Florian Wolters  wrote:

> Hi @ll,
>
> I just moved my Asterisk Box and changed the Provider and Internet Access
> to a full IP Access by Deutsche Telekom.
>
> I set up my sip.conf as I found various examples throughout the Net. Calls
> and some other stuff is basically working.
>
> The problem I ran into is, that the outgoing and incoming calls are
> dropped after exactly 15 Minutes. Solution for this should be setting the
> session-timers to refuse but this doesnt change anything here.
>
> I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest
> Asterisk by Digium without success.
>
> Has anyone else has the Same problem or is a solution already known? Could
> someone point me in the right direction? I can provide (debug) logs if
> essential.
>
> Best regards
>
>Flo
>
>
> --
> _
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>http://www.asterisk.org/hello
>
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>



-- 
Rob Krakora
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677 Ext 212
(317)663-0808 Fax
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Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

2013-03-21 Thread Florian Wolters

Hello,

> I solved it by moving Asterisk 1.6 to Asterisk 1.4.
>
> Try asterisk 1.4 or 1.8  on a test box and see how it goes.

I did try the latest 1.8.2x release already without any improvement.
Currently running is a Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 as the tcpdump
says (little mistake to my last mail).

I also played around with "canreinvite". But regardless of the setting
(yes/no) I still get disconnects after 15 minutes. I just tried to accept
session-timers, but this has no connection to this issue either.

So I turned on SIP debug for this host and analyszed it with wireshark.
The last packets show an INVITE from my provider, that is answered by my
Asterisk with "200 OK, with session description". What follows is an ACK
by the provider and immediately a BYE sent by the provider. So for me it
looks like the provider is disconnecting the call.

I could not see any reason or hangup cause for this in the dump. Are there
error messages for this that can be seen in the protocol?

The tcpdump (the last few packets) shows:


--- 8< snip ---

13:37:28.258566 IP (tos 0x0, ttl 64, id 44187, offset 0, flags [DF], proto
TCP (6), length 611)
172.16.0.2.44929 > 217.0.17.170.5060: Flags [P.], cksum 0xf764
(incorrect -> 0xd1be), seq 4568:5139, ack 4057, win 45600, length 571
13:37:28.277390 IP (tos 0xc0, ttl 55, id 4807, offset 0, flags [DF], proto
TCP (6), length 547)
217.0.17.170.5060 > 172.16.0.2.44929: Flags [P.], cksum 0x2c63
(correct), seq 4057:4564, ack 5139, win 65535, length 507
13:37:28.277415 IP (tos 0x0, ttl 64, id 44188, offset 0, flags [DF], proto
TCP (6), length 40)
172.16.0.2.44929 > 217.0.17.170.5060: Flags [.], cksum 0xf529
(incorrect -> 0xdc6d), ack 4564, win 45600, length 0
13:37:54.240304 IP (tos 0xc0, ttl 25, id 14090, offset 0, flags [none],
proto UDP (17), length 1255)
217.0.17.170.5060 > 172.16.0.2.5060: SIP, length: 1227
INVITE sip:090066@79.253.136.104:5060 SIP/2.0
Via: SIP/2.0/UDP
217.0.17.170:5060;branch=z9hG4bK354b7c6fb4f56e91ecb6a22cd865b03b.0e83c94b
Via: SIP/2.0/TCP
62.156.82.55:5060;branch=z9hG4bK88ecd3ab295c71dfc9ea9ab6c3134b08jaaiaaj23clqa3Zqkv7akae3e3wetjnxm
Via: SIP/2.0/TCP 62.156.80.48:5083;branch=z9hG4bK2609300495-692831843
Max-Forwards: 70
To: ;tag=as77f2fb84
From: ;tag=8f233b97
Call-ID: 83de2b0c3faf0ef9@217.0.17.170
Contact:
;+g.3gpp.icsi-ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel";+g.3gpp.icsi_ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel"
CSeq: 1939619 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER,
SUBSCRIBE, UPDATE
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 297

v=0
o=- 558131575 1701401067 IN IP4 217.0.17.170
s=Phone Call via hiQ9200 SIPCA
c=IN IP4 217.0.1.67
t=0 0
m=audio 16884 RTP/AVP 8 100
b=AS:110
b=RS:1375
b=RR:4125
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sqn: 0
a=sendrecv
a=ptime:20

13:37:54.240497 IP (tos 0xc0, ttl 25, id 14091, offset 0, flags [none],
proto UDP (17), length 1222)
217.0.17.170.5060 > 172.16.0.2.5060: SIP, length: 1194
INVITE sip:090066@79.253.136.104:5060;transport=TCP SIP/2.0
Via: SIP/2.0/UDP
217.0.17.170:5060;branch=z9hG4bK86e8a99ff6e15f3e72053880758bf877.9856a4e1
Via: SIP/2.0/TCP
62.156.82.55:5060;branch=z9hG4bKb0d91d40f6f840a4ff61c11c2afe12d6jaaiaahr0zo2a3Zqkv7awon0rib4uosfa
Via: SIP/2.0/TCP 62.156.80.48:5082;branch=z9hG4bK2609108269-1839751709
Max-Forwards: 70
To: 090066 ;tag=as09bca4fd
From: ;tag=f18b4044
Call-ID: 248ef1b5553e5756490d6556573a1...@tel.t-online.de
Contact:
;+g.3gpp.icsi-ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel";+g.3gpp.icsi_ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel"
CSeq: 1939639 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, PUBLISH, REFER,
REGISTER, SUBSCRIBE, UPDATE
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 224

v=0
o=- 1028575251 1704720679 IN IP4 217.0.17.170
s=Basic Session
c=IN IP4 217.0.1.81
t=0 0
m=audio 17120 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

13:37:54.240593 IP (tos 0x0, ttl 64, id 43415, offset 0, flags [none],
proto UDP (17), length 782)
172.16.0.2.5060 > 217.0.17.170.5060: SIP, length: 754
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
217.0.17.170:5060;branch=z9hG4bK354b7c6fb4f56e91ecb6a22cd865b03b.0e83c94b;received=217.0.17.170;rport=5060
Via: SIP/2.0/TCP
62.156.82.55:5060;branch=z9hG4bK88ecd3ab295c71dfc9ea9ab6c3134b08jaaiaaj23clqa3Zqkv7akae3e3wetjnxm

Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

2013-03-21 Thread Matthew J. Roth

Florian Wolters wrote:
> 
> So I turned on SIP debug for this host and analyszed it with wireshark.
> The last packets show an INVITE from my provider, that is answered by my
> Asterisk with "200 OK, with session description". What follows is an ACK
> by the provider and immediately a BYE sent by the provider. So for me it
> looks like the provider is disconnecting the call.

Florian,

This is a little hard to diagnose without seeing the SIP traffic for the
duration of the call.  It makes it impossible to tell if the INVITES the
provider is sending are related to the call (i.e. have the same Call-ID header),
but if they are being sent consistently 15 minutes into every call it may not
matter.  If the provider is sending you unsolicited INVITES that cause your
calls to drop, I'd suggest contacting their customer service and asking them why
they are being sent.

The provider actually sent you two INVITES in rapid succession with different
Call-IDs.  To keep this simple, I'll use the following shorthand:

  Call 1 = Call-ID 83de2b0c3faf0ef9@217.0.17.170
  Call 2 = Call-ID 248ef1b5553e5756490d6556573a1...@tel.t-online.de

Call 1 is terminated with a BYE from Asterisk immediately after it gets the ACK
from the provider.  The provider tried to terminate it immediately with its own
BYE, but it lost the race.  This results in the Call/Transaction Does Not Exist
message at the end of that dialog.

Call 2 is terminated with a BYE from the provider immediately after they ACK the
OK from Asterisk.

As I said above, I'd start out by asking the provider why they are sending these
INVITES in the first place.  Here is the simple timeline derived from your SIP
trace that I worked from:

  Call 1  Call 2

  13:37:54.240304
INVITE From Provider to Asterisk

  13:37:54.240497
INVITE From Provider to Asterisk

  13:37:54.240593
Trying From Asterisk to Provider

  13:37:54.240752
OK From Asterisk to Provider

  13:37:54.240976
Trying From Asterisk to Provider

  13:37:54.241172
OK From Asterisk to Provider

  13:37:54.282723
ACK From Provider to Asterisk

  13:37:54.286434
BYE From Provider to Asterisk

  13:37:54.286700
OK From Asterisk to Provider

  13:37:54.339838
OK From Asterisk to Provider

  13:37:54.384756
ACK From Provider to Asterisk

  13:37:54.385007
BYE From Asterisk to Provider

  13:37:54.388625
BYE From Provider to Asterisk

  13:37:54.388816
OK From Asterisk to Provider

  13:37:54.404027
Call/Transaction Does Not Exist From
Provider to Asterisk

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] Diagnosing call problem

2013-03-21 Thread Matthew J. Roth

Mitch Claborn wrote:
> 
> Thank you for that most excellent post.  I had guessed at most of the 
> SDP fields and meaning.

No problem.  I actually like looking at SIP traces for some reason.

> I have wireshark traces from the client and the RTP packets are not in 
> the trace, which I think means that the client software is simply not 
> producing them.  I have opened a ticket with SFL phone support and will 
> post here if I find anything.

That's a reasonable conclusion.  Just make sure that you get some traces of good
calls to verify that your tests are valid.

> I did test the "muted microphone" theory.  SFLphone continues to send 
> RTP packets even when the mic is muted, so that doesn't seem to be the 
> cause.

It's always a good idea to rule out PEBKAC before spending a lot of time
diagnosing a problem.

> I've also compared the call initiation SIP and SDP packets between a 
> call that fails and one that works correctly.  I can discern no 
> difference other than things like port numbers and call IDs.
> 
> Tomorrow I'll be trying one of my agents on Bria instead of SFL - maybe 
> that will make a difference.

It really seems like it may be a problem with the softphone.  I'm sure the
developers of SFLphone will appreciate your feedback, because not sending RTP is
a pretty serious bug.

I'll keep an eye on this thread and help out if I can.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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[asterisk-users] Need help about round-robin

2013-03-21 Thread Salaheddine Elharit

hello list,

i have installed 2 diguim cards in my server using asterisk 1.4 (i use the
old version with zapata.conf and zaptel.conf)

i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i
want to active the round-robin for span 2 and 6) in order to activate the
WIMAX and FH

please see the configuration below and tell me if there is anything  wrong

question 2: what is difference between etc\zapataa.conf and
etc\asterisk\zapata.conf

i make this configuration just in etc\asterisk\zapata.conf i don't know if
i must do this configuration also in etc\zapata.conf

etc\asterisk\zapata.conf


[channels]
context=default
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
rxgain=0.0
txgain=0.0

group=1
switchtype=euroisdn
signalling=pri_cpe
callgroup=1
pickupgroup=1
immediate=no
channel => 1-15,17-31

group=2
callgroup=2
switchtype=qsig
signalling=pri_net
callerid=X(my callerID)
immediate=no
channel => 156-170
channel => 172-176
channel => 32-46
channel => 48-62


etc\zaptel.conf

# Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not
hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS RED
span=1,1,0,ccs,hdb3
# termtype: te
bchan=1-15,17-31
dchan=16

# Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" HDB3/CCS RED
span=2,2,0,ccs,hdb3
# termtype: te
bchan=32-46,48-62
dchan=47

# Span 3: TE4/0/3 "T4XXP (PCI) Card 0 Span 3"
# span=3,3,0,ccs,hdb3
# termtype: te
# bchan=63-77,79-93
# dchan=78

# Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4"
# span=4,4,0,ccs,hdb3
# termtype: te
# bchan=94-108,110-124
# dchan=109

# Span 5: TE2/1/1 "T2XXP (PCI) Card 1 Span 1"
#span=5,5,0,ccs,hdb3
# termtype: te
#bchan=125-139,141-155
#dchan=140

# Span 6: TE2/1/2 "T2XXP (PCI) Card 1 Span 2"
span=6,6,0,ccs,hdb3
# termtype: te
bchan=156-170,172-186
dchan=171

# Global data

loadzone = us
defaultzone = us

thank you so much
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Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

2013-03-21 Thread Jim Lucas


On 3/21/2013 12:31 AM, Florian Wolters wrote:

Hi @ll,

I just moved my Asterisk Box and changed the Provider and Internet Access to a 
full IP Access by Deutsche Telekom.

I set up my sip.conf as I found various examples throughout the Net. Calls and 
some other stuff is basically working.

The problem I ran into is, that the outgoing and incoming calls are dropped 
after exactly 15 Minutes. Solution for this should be setting the 
session-timers to refuse but this doesnt change anything here.

I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest 
Asterisk by Digium without success.

Has anyone else has the Same problem or is a solution already known? Could 
someone point me in the right direction? I can provide (debug) logs if 
essential.

Best regards

Flo


Florian,

As both an VoIP provider and phone system vendor, I had this same 
problem 2 years ago.  In my situation, it turned out that it was nothing 
to do with either the Asterisk box or the provider.


The problem was with a router that we had terminating our T1 connection. 
 As an ISP we provide T1's to many customers and we provide the router 
as well.  In this specific case, the customer purchased a data T1 
connection with QoS (sip and rtp) then purchased our IP asterisk phone 
system with SIP trunks from us as well.


The way we found this issue was by switching our the T1 router.  Turns 
out that it fixed the problem.  Exact same configuration was on each 
router.  So we started scratching our heads...


We then looked at the firmware of the two routers and found that they 
were different.


We provide Cisco 26XX routers.

Their are many places on the net talking about the 15 minute NAT timeout 
issue.


If you are not using this device, well, maybe it has a similar bug.

--
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Re: [asterisk-users] Need help about round-robin

2013-03-21 Thread Bharat Lalcheta

What do you mean by roundrobin here
On Mar 21, 2013 8:27 PM, "Salaheddine Elharit" 
wrote:

> hello list,
>
> i have installed 2 diguim cards in my server using asterisk 1.4 (i use the
> old version with zapata.conf and zaptel.conf)
>
> i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i
> want to active the round-robin for span 2 and 6) in order to activate the
> WIMAX and FH
>
> please see the configuration below and tell me if there is anything  wrong
>
> question 2: what is difference between etc\zapataa.conf and
> etc\asterisk\zapata.conf
>
> i make this configuration just in etc\asterisk\zapata.conf i don't know if
> i must do this configuration also in etc\zapata.conf
>
> etc\asterisk\zapata.conf
>
>
> [channels]
> context=default
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> rxgain=0.0
> txgain=0.0
>
> group=1
> switchtype=euroisdn
> signalling=pri_cpe
> callgroup=1
> pickupgroup=1
> immediate=no
> channel => 1-15,17-31
>
> group=2
> callgroup=2
> switchtype=qsig
> signalling=pri_net
> callerid=X(my callerID)
> immediate=no
> channel => 156-170
> channel => 172-176
> channel => 32-46
> channel => 48-62
>
>
> etc\zaptel.conf
>
> # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not
> hand edit
> # Zaptel Configuration File
> #
> # This file is parsed by the Zaptel Configurator, ztcfg
> #
> # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS RED
> span=1,1,0,ccs,hdb3
> # termtype: te
> bchan=1-15,17-31
> dchan=16
>
> # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" HDB3/CCS RED
> span=2,2,0,ccs,hdb3
> # termtype: te
> bchan=32-46,48-62
> dchan=47
>
> # Span 3: TE4/0/3 "T4XXP (PCI) Card 0 Span 3"
> # span=3,3,0,ccs,hdb3
> # termtype: te
> # bchan=63-77,79-93
> # dchan=78
>
> # Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4"
> # span=4,4,0,ccs,hdb3
> # termtype: te
> # bchan=94-108,110-124
> # dchan=109
>
> # Span 5: TE2/1/1 "T2XXP (PCI) Card 1 Span 1"
> #span=5,5,0,ccs,hdb3
> # termtype: te
> #bchan=125-139,141-155
> #dchan=140
>
> # Span 6: TE2/1/2 "T2XXP (PCI) Card 1 Span 2"
> span=6,6,0,ccs,hdb3
> # termtype: te
> bchan=156-170,172-186
> dchan=171
>
> # Global data
>
> loadzone = us
> defaultzone = us
>
> thank you so much
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
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Re: [asterisk-users] Need help about round-robin

2013-03-21 Thread Salaheddine Elharit

i mean the burden-sharing between Wimax and FH

2013/3/21 Bharat Lalcheta 

> What do you mean by roundrobin here
> On Mar 21, 2013 8:27 PM, "Salaheddine Elharit" 
> wrote:
>
>> hello list,
>>
>> i have installed 2 diguim cards in my server using asterisk 1.4 (i use
>> the old version with zapata.conf and zaptel.conf)
>>
>> i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i
>> want to active the round-robin for span 2 and 6) in order to activate the
>> WIMAX and FH
>>
>> please see the configuration below and tell me if there is anything  wrong
>>
>> question 2: what is difference between etc\zapataa.conf and
>> etc\asterisk\zapata.conf
>>
>> i make this configuration just in etc\asterisk\zapata.conf i don't know
>> if i must do this configuration also in etc\zapata.conf
>>
>> etc\asterisk\zapata.conf
>>
>>
>> [channels]
>> context=default
>> hidecallerid=no
>> callwaiting=yes
>> usecallingpres=yes
>> callwaitingcallerid=yes
>> threewaycalling=yes
>> transfer=yes
>> canpark=yes
>> cancallforward=yes
>> callreturn=yes
>> rxgain=0.0
>> txgain=0.0
>>
>> group=1
>> switchtype=euroisdn
>> signalling=pri_cpe
>> callgroup=1
>> pickupgroup=1
>> immediate=no
>> channel => 1-15,17-31
>>
>> group=2
>> callgroup=2
>> switchtype=qsig
>> signalling=pri_net
>> callerid=X(my callerID)
>> immediate=no
>> channel => 156-170
>> channel => 172-176
>> channel => 32-46
>> channel => 48-62
>>
>>
>> etc\zaptel.conf
>>
>> # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do
>> not hand edit
>> # Zaptel Configuration File
>> #
>> # This file is parsed by the Zaptel Configurator, ztcfg
>> #
>> # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS RED
>> span=1,1,0,ccs,hdb3
>> # termtype: te
>> bchan=1-15,17-31
>> dchan=16
>>
>> # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" HDB3/CCS RED
>> span=2,2,0,ccs,hdb3
>>  # termtype: te
>> bchan=32-46,48-62
>> dchan=47
>>
>> # Span 3: TE4/0/3 "T4XXP (PCI) Card 0 Span 3"
>> # span=3,3,0,ccs,hdb3
>> # termtype: te
>> # bchan=63-77,79-93
>> # dchan=78
>>
>> # Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4"
>> # span=4,4,0,ccs,hdb3
>> # termtype: te
>> # bchan=94-108,110-124
>> # dchan=109
>>
>> # Span 5: TE2/1/1 "T2XXP (PCI) Card 1 Span 1"
>> #span=5,5,0,ccs,hdb3
>> # termtype: te
>> #bchan=125-139,141-155
>> #dchan=140
>>
>> # Span 6: TE2/1/2 "T2XXP (PCI) Card 1 Span 2"
>> span=6,6,0,ccs,hdb3
>> # termtype: te
>> bchan=156-170,172-186
>> dchan=171
>>
>> # Global data
>>
>> loadzone = us
>> defaultzone = us
>>
>> thank you so much
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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Re: [asterisk-users] Need help about round-robin

2013-03-21 Thread Bharat Lalcheta

If u want to dial in round robin use Dial(zap/r2/2) . It dials using
channel in round robin
On Mar 21, 2013 9:37 PM, "Salaheddine Elharit" 
wrote:

> i mean the burden-sharing between Wimax and FH
>
> 2013/3/21 Bharat Lalcheta 
>
>> What do you mean by roundrobin here
>> On Mar 21, 2013 8:27 PM, "Salaheddine Elharit" <
>> salah.elharit...@gmail.com> wrote:
>>
>>> hello list,
>>>
>>> i have installed 2 diguim cards in my server using asterisk 1.4 (i use
>>> the old version with zapata.conf and zaptel.conf)
>>>
>>> i want to use the span 1 for group 1 and span 2-span 6 for the group 2
>>> (i want to active the round-robin for span 2 and 6) in order to activate
>>> the WIMAX and FH
>>>
>>> please see the configuration below and tell me if there is anything
>>>  wrong
>>>
>>> question 2: what is difference between etc\zapataa.conf and
>>> etc\asterisk\zapata.conf
>>>
>>> i make this configuration just in etc\asterisk\zapata.conf i don't know
>>> if i must do this configuration also in etc\zapata.conf
>>>
>>> etc\asterisk\zapata.conf
>>>
>>>
>>> [channels]
>>> context=default
>>> hidecallerid=no
>>> callwaiting=yes
>>> usecallingpres=yes
>>> callwaitingcallerid=yes
>>> threewaycalling=yes
>>> transfer=yes
>>> canpark=yes
>>> cancallforward=yes
>>> callreturn=yes
>>> rxgain=0.0
>>> txgain=0.0
>>>
>>> group=1
>>> switchtype=euroisdn
>>> signalling=pri_cpe
>>> callgroup=1
>>> pickupgroup=1
>>> immediate=no
>>> channel => 1-15,17-31
>>>
>>> group=2
>>> callgroup=2
>>> switchtype=qsig
>>> signalling=pri_net
>>> callerid=X(my callerID)
>>> immediate=no
>>> channel => 156-170
>>> channel => 172-176
>>> channel => 32-46
>>> channel => 48-62
>>>
>>>
>>> etc\zaptel.conf
>>>
>>> # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do
>>> not hand edit
>>> # Zaptel Configuration File
>>> #
>>> # This file is parsed by the Zaptel Configurator, ztcfg
>>> #
>>> # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS RED
>>> span=1,1,0,ccs,hdb3
>>> # termtype: te
>>> bchan=1-15,17-31
>>> dchan=16
>>>
>>> # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" HDB3/CCS RED
>>> span=2,2,0,ccs,hdb3
>>>  # termtype: te
>>> bchan=32-46,48-62
>>> dchan=47
>>>
>>> # Span 3: TE4/0/3 "T4XXP (PCI) Card 0 Span 3"
>>> # span=3,3,0,ccs,hdb3
>>> # termtype: te
>>> # bchan=63-77,79-93
>>> # dchan=78
>>>
>>> # Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4"
>>> # span=4,4,0,ccs,hdb3
>>> # termtype: te
>>> # bchan=94-108,110-124
>>> # dchan=109
>>>
>>> # Span 5: TE2/1/1 "T2XXP (PCI) Card 1 Span 1"
>>> #span=5,5,0,ccs,hdb3
>>> # termtype: te
>>> #bchan=125-139,141-155
>>> #dchan=140
>>>
>>> # Span 6: TE2/1/2 "T2XXP (PCI) Card 1 Span 2"
>>> span=6,6,0,ccs,hdb3
>>> # termtype: te
>>> bchan=156-170,172-186
>>> dchan=171
>>>
>>> # Global data
>>>
>>> loadzone = us
>>> defaultzone = us
>>>
>>> thank you so much
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>
>
>
> --
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Re: [asterisk-users] Need help about round-robin

2013-03-21 Thread Steve Edwards


On Thu, 21 Mar 2013, Salaheddine Elharit wrote:

i have installed 2 diguim cards in my server using asterisk 1.4 (i use 
the old version with zapata.conf and zaptel.conf)


question 2: what is difference between etc\zapataa.conf and 
etc\asterisk\zapata.conf


There is no /etc/zapata.conf.

The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.

Note that the direction of the 'slash' is significant as is the leading 
slash.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Need help about round-robin

2013-03-21 Thread Bharat Lalcheta

File is ok there is no etc/zapata file.
On Mar 21, 2013 9:42 PM, "Steve Edwards"  wrote:

> On Thu, 21 Mar 2013, Salaheddine Elharit wrote:
>
>  i have installed 2 diguim cards in my server using asterisk 1.4 (i use
>> the old version with zapata.conf and zaptel.conf)
>>
>> question 2: what is difference between etc\zapataa.conf and
>> etc\asterisk\zapata.conf
>>
>
> There is no /etc/zapata.conf.
>
> The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.
>
> Note that the direction of the 'slash' is significant as is the leading
> slash.
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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>   
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>
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Re: [asterisk-users] SIP account registration fails after upgrade to 1.8

2013-03-21 Thread Jaap Winius

On Tue, 19 Mar 2013 02:15:10 +, Jaap Winius wrote:

> Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9
> to 1.8.13, my server is no longer able to register a connection to a SIP
> account at my ISP (XS4ALL in the Netherlands). At the same time, it is
> still able to register a different account with another SIP provider...

To answer my own question, this turned out to be due to a bug in the SIP 
server at XS4ALL. I discovered it after using tcpdump to examine the 
exchange of packets during my registration attempts and noticing that 
Asterisk 1.8.13.1 was using an IPv6 address in the Call-ID instead of an 
IPv4 address as before. According to the specification for SIP 2.0 (RFC 
3261) this is perfectly legal, just as long as both parties treat the 
entire Call-ID as a string and never make any changes to it.

However, I discovered that is was exactly what the SIP server at XS4ALL 
is doing. For example, if my server sends it a SIP packet with a register 
request and a Call-ID that looks like this:

   Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:abcd:1::a]

... somewhere along they line they end up changing it to this:

   Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:ABCD:1::A]

In other words, it is treating the latter part of the Call-ID not as a 
string, but as an IPv6 address and has taken it upon itself to change all 
of the letters in that address to upper case. This changes the Call-ID 
and thus my registration attemp cannot be completed. Of course, this 
won't affect you if you happen to have an IPv6 address without any 
letters in it.

This situation is in contrast to another SIP provider that I use, 
sip.internetcalls.com, with which I currently have no problems because 
they leave such Call-IDs unchanged. I don't know what kind of SIP server 
software they use, but XS4ALL appears to be using Cirpack 4.42a.

This bug is very similar to another one described in this forum exchange:

   http://forums.asterisk.org/viewtopic.php?f=1&t=84603&start=0

Here, a SIP server at an ISP was taking the IPv6 address at the end of a 
Call-ID and expanding it, e.g. from ::1 (the IPv6 loopback address) to 
0:0:0:0:0:0:0:1. In both that case and in mine, we get the same result: 
an altered Call-ID that leads to endless timeouts and no registration.

Hopefully, my ISP will see fit to squash this bug ASAP.

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[asterisk-users] Asterisk 1.8 and dual stack support

2013-03-21 Thread Jaap Winius

Hi folks,

Following an upgrade to Debian wheezy, I'm now running Asterisk 1.8.13.1. 
As opposed to Asterisk 1.6.2.9 that I ran with squeeze, this version can 
support IPv6. However, it seems that I can't get it to support both IPv4 
and IPv6 at the same time. For example, if in sip.conf I set the bindaddr 
variable to '::' it will only listen on IPv6 and none of my IPv4-only 
friends and peers will be able to connect to it. On the other hand, if I 
set it to '0.0.0.0' then it will not listen on IPv6.

Is this a bug, or is this simply a limitation of Asterisk 1.8.13.1, or is 
there some other way to configure it for dual-stack support?

Thanks,

Jaap


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Re: [asterisk-users] Asterisk 1.8 and dual stack support

2013-03-21 Thread Asghar Mohammad

please see,

http://lists.digium.com/pipermail/asterisk-users/2013-March/278130.html

On Thu, Mar 21, 2013 at 5:47 PM, Jaap Winius  wrote:

> Hi folks,
>
> Following an upgrade to Debian wheezy, I'm now running Asterisk 1.8.13.1.
> As opposed to Asterisk 1.6.2.9 that I ran with squeeze, this version can
> support IPv6. However, it seems that I can't get it to support both IPv4
> and IPv6 at the same time. For example, if in sip.conf I set the bindaddr
> variable to '::' it will only listen on IPv6 and none of my IPv4-only
> friends and peers will be able to connect to it. On the other hand, if I
> set it to '0.0.0.0' then it will not listen on IPv6.
>
> Is this a bug, or is this simply a limitation of Asterisk 1.8.13.1, or is
> there some other way to configure it for dual-stack support?
>
> Thanks,
>
> Jaap
>
>
> --
> _
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Re: [asterisk-users] Need help about round-robin

2013-03-21 Thread Salaheddine Elharit

how can i use Dial(zap/r2/2)

below an exemple from my extensions.conf

exten => _0612.,1,Set(CALLERID(number)=520460587)
exten => _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =>
_0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten => _0612.,n,Hangup();

thanks and regards.

2013/3/21 Bharat Lalcheta 

> File is ok there is no etc/zapata file.
> On Mar 21, 2013 9:42 PM, "Steve Edwards" 
> wrote:
>
>> On Thu, 21 Mar 2013, Salaheddine Elharit wrote:
>>
>>  i have installed 2 diguim cards in my server using asterisk 1.4 (i use
>>> the old version with zapata.conf and zaptel.conf)
>>>
>>> question 2: what is difference between etc\zapataa.conf and
>>> etc\asterisk\zapata.conf
>>>
>>
>> There is no /etc/zapata.conf.
>>
>> The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.
>>
>> Note that the direction of the 'slash' is significant as is the leading
>> slash.
>>
>> --
>> Thanks in advance,
>> --**--**
>> -
>> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>> Newline  Fax: +1-760-731-3000
>>
>> --
>> __**__**_
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   
>> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>>
>
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Re: [asterisk-users] Need help about round-robin

2013-03-21 Thread Bharat Lalcheta

Use r2 instead of g2 in dial

Dial(Zap/r2/${EXTEN}
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Re: [asterisk-users] Need help about round-robin

2013-03-21 Thread Asghar Mohammad

hi,

exten => _0612.,1,Set(CALLERID(number)=520460587)
exten => _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =>
_0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten => _0612.,n,Hangup()

Note r in Dial.
you can use r for Ascending and R for Descending order

On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit <
salah.elharit...@gmail.com> wrote:

> how can i use Dial(zap/r2/2)
>
> below an exemple from my extensions.conf
>
> exten => _0612.,1,Set(CALLERID(number)=520460587)
> exten => _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
> exten => _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
> exten =>
> _0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
> exten => _0612.,n,Hangup();
>
> thanks and regards.
>
> 2013/3/21 Bharat Lalcheta 
>
>> File is ok there is no etc/zapata file.
>> On Mar 21, 2013 9:42 PM, "Steve Edwards" 
>> wrote:
>>
>>> On Thu, 21 Mar 2013, Salaheddine Elharit wrote:
>>>
>>>  i have installed 2 diguim cards in my server using asterisk 1.4 (i use
 the old version with zapata.conf and zaptel.conf)

 question 2: what is difference between etc\zapataa.conf and
 etc\asterisk\zapata.conf

>>>
>>> There is no /etc/zapata.conf.
>>>
>>> The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.
>>>
>>> Note that the direction of the 'slash' is significant as is the leading
>>> slash.
>>>
>>> --
>>> Thanks in advance,
>>> --**--**
>>> -
>>> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867PST
>>> Newline  Fax:
>>> +1-760-731-3000
>>>
>>> --
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Re: [asterisk-users] Diagnosing call problem

2013-03-21 Thread Mitch Claborn


I did open a ticket with SFL support and sent them the packet trace.

Interestingly, using Bria we sometimes see similar, though not exactly 
the same, symptoms.  That would make me wonder about the TCP stack on 
the client machine, or similar.


Bria on Ubuntu is not terribly stable.  Bria on the Mac works very well, 
but that's a pretty expensive solution.


We are close to ditching the soft phones entirely for this call center 
and going to the Digium D40.  I put one of those in service this morning 
and the calls are noticeably clearer and there have been no reported 
problems.



Mitch

On 03/21/2013 09:48 AM, Matthew J. Roth wrote:

Mitch Claborn wrote:


Thank you for that most excellent post.  I had guessed at most of the
SDP fields and meaning.


No problem.  I actually like looking at SIP traces for some reason.


I have wireshark traces from the client and the RTP packets are not in
the trace, which I think means that the client software is simply not
producing them.  I have opened a ticket with SFL phone support and will
post here if I find anything.


That's a reasonable conclusion.  Just make sure that you get some traces of good
calls to verify that your tests are valid.


I did test the "muted microphone" theory.  SFLphone continues to send
RTP packets even when the mic is muted, so that doesn't seem to be the
cause.


It's always a good idea to rule out PEBKAC before spending a lot of time
diagnosing a problem.


I've also compared the call initiation SIP and SDP packets between a
call that fails and one that works correctly.  I can discern no
difference other than things like port numbers and call IDs.

Tomorrow I'll be trying one of my agents on Bria instead of SFL - maybe
that will make a difference.


It really seems like it may be a problem with the softphone.  I'm sure the
developers of SFLphone will appreciate your feedback, because not sending RTP is
a pretty serious bug.

I'll keep an eye on this thread and help out if I can.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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[asterisk-users] Howto create variable from the name of another one and get content of it

2013-03-21 Thread Administrator TOOTAI


Hello,

I have a variable created like

... Set(__myVar=${ARG1})
... Set(__${myVar}STATUS=)

If ARG1 is abcd, variable is abcdSTATUS and should be empty. This is OK.

Now I would like to get the value of abcdSTATUS. How to do it? 
${${myVar}STATUS}} isn't working, nor ${{myvar}STATUS}


Thanks for any hint

--
Daniel

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[asterisk-users] Cisco SPA 5xx/3xx/9xx phones don't respond to SIPAddHeader(Call-Info: answer-after=0)

2013-03-21 Thread Carlos Alvarez

All other phones we work with will auto-answer when we do this:

[macro-paging1way]
exten => s,1,SIPAddHeader(Call-Info: answer-after=0)
exten => s,n,Page(${PAGINGLIST})
exten => s,n, Hangup

The SPA phones simply ring.  I have verified that Auto Answer Page is set
to yes (the default).  We've tried a variety of firmware versions and phone
ages, going back to an old 942 and new 504s.  Any ideas?

-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Cisco SPA 5xx/3xx/9xx phones don't respond to SIPAddHeader(Call-Info: answer-after=0)

2013-03-21 Thread Optical Phoenix

On Thu, Mar 21, 2013 at 2:48 PM, Carlos Alvarez wrote:

> All other phones we work with will auto-answer when we do this:
>
> [macro-paging1way]
> exten => s,1,SIPAddHeader(Call-Info: answer-after=0)
> exten => s,n,Page(${PAGINGLIST})
> exten => s,n, Hangup
>
> The SPA phones simply ring.  I have verified that Auto Answer Page is set
> to yes (the default).  We've tried a variety of firmware versions and phone
> ages, going back to an old 942 and new 504s.  Any ideas?
>
> --
> Carlos Alvarez
> TelEvolve
> 602-889-3003
>
>
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Hi Carlos,
According to this site,
http://community.linksys.com/t5/VoIP-Phones/SPA942-auto-answer-page-intercom-beeping-loudly/td-p/215064the
sip string should be "
Call-Info:\;answer-after=0". I have not tested this yet however.
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Re: [asterisk-users] Howto create variable from the name of another one and get content of it

2013-03-21 Thread Asghar Mohammad

hi,
${myVar}STATUS is empty you have not assign any value here your var
Set(__${myVar}STATUS=) is empty.
use instead  Set(__myVar=${ARG1}STATUS) and remove second line.

On Thu, Mar 21, 2013 at 7:45 PM, Administrator TOOTAI wrote:

> Hello,
>
> I have a variable created like
>
> ... Set(__myVar=${ARG1})
> ... Set(__${myVar}STATUS=)
>
> If ARG1 is abcd, variable is abcdSTATUS and should be empty. This is OK.
>
> Now I would like to get the value of abcdSTATUS. How to do it?
> ${${myVar}STATUS}} isn't working, nor ${{myvar}STATUS}
>
> Thanks for any hint
>
> --
> Daniel
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Allow/Disallow

2013-03-21 Thread Nick Khamis

Hello Everyone,

I have disallow=all and allow=g729 set in sip.conf however, it seems
that asterisk still thinks it support other codecs:

Capabilities: us - 0x8008000e (gsm|ulaw|alaw|h263|testlaw). How
can I disable gsm,ulaw,alaw.

Thanks in Advance,

Nick.

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Re: [asterisk-users] Cisco SPA 5xx/3xx/9xx phones don't respond to SIPAddHeader(Call-Info: answer-after=0)

2013-03-21 Thread Carlos Alvarez

On Thu, Mar 21, 2013 at 11:58 AM, Optical Phoenix
wrote:

>
>
> Hi Carlos,
> According to this site,
> http://community.linksys.com/t5/VoIP-Phones/SPA942-auto-answer-page-intercom-beeping-loudly/td-p/215064the
>  sip string should be "
> Call-Info:\;answer-after=0". I have not tested this yet however.
>

Thanks, that does work.  Seems to not interfere with the Grandstream and
Polycom phones' operation either.

-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Allow/Disallow

2013-03-21 Thread Asghar Mohammad

please post sip.conf.

On Thu, Mar 21, 2013 at 8:01 PM, Nick Khamis  wrote:

> Hello Everyone,
>
> I have disallow=all and allow=g729 set in sip.conf however, it seems
> that asterisk still thinks it support other codecs:
>
> Capabilities: us - 0x8008000e (gsm|ulaw|alaw|h263|testlaw). How
> can I disable gsm,ulaw,alaw.
>
> Thanks in Advance,
>
> Nick.
>
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Re: [asterisk-users] Delay before audio starts

2013-03-21 Thread Gerard

> I think a simple tcpdump of the traffic will show the mystery. It can
> be your provider doing something nasty. Have you tried using some
> other cheap SIP termination? or arrange a fake termination yourself
> on another server?
> 
> Leandro

I thought so too, but it doesn't appear to .

I just bought a door intercom device, set up the extension for it and
it's doing the same thing, when it connects there is a 10 second delay
before the other side can hear my voice.
However watching tcpdump, the audio starts streaming both ways immediately.
Changing the dialplan fixes the issue:
957 => { // Test door phone
Answer(); //  <--- this line fixes the problem!
Dial(SIP/199,20);
Hangup();
};

It's an ok workaround for the door intercom, but in the case of the
forwarded calls below, as soon as I Answer() their ringback disappears
and the line goes dead while they wait for our guy to answer the phone.

I may start a separate post about getting ringback to work after Answer();

Thanks for the help by the way.
-Gerard

On 03/01/13 14:34, Leandro Dardini wrote:

> 
> 2013/3/1 Gerard 
> 
>> I thought it was the re-invites too, but I have it turned off
>> everywhere.
>> 
>> On 03/01/13 08:36, Eric Wieling wrote:
>>> When Answer fixes the issue, the root cause is often NAT (could
>>> be
>> firewall) since Answering the call prevents any reinvites.
>>> 
>>> -Original Message- From:
>>> asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
>>> Sent: Friday, March 01, 2013 9:33 AM To:
>>> asterisk-users@lists.digium.com Subject: Re: [asterisk-users]
>>> Delay before audio starts
>>> 
>>> I've found a workaround of sorts, If I change my below code to : 
>>> 1AA => { NoOp(${CALLERID(num)}); Answer();  //
>>> <--- add this Ringing; 
>>> Set(CHANNEL(musicclass)=none); 
>>> Dial(${OUTBOUND-TRUNKR}/1XX,30); Voicemail(198,u); };
>>> 
>>> That fixes the issue. It doesn't fix the call forward issue on
>>> the phone
>> though. I've made a few extra extensions, one each corresponding to
>> a number he wants to call forward to, if I have him forward to the
>> extensions who then forward to the real number, it works, thanks to
>> adding "Answer()" to the dialplan.
>>> 
>>> -Gerard
>>> 
>>> 
>>> On 02/26/13 13:19, Gerard wrote:
 Hi everyone,

 I'm having a hard time figuring this issue out, we just
 switched from a T1 PRI to a SIP trunk provider and that's when
 the issue started. Now when someone forwards all calls on their
 phone to a cellphone, when a customer calls in, Asterisk
 correctly calls the cellphone and connects the call, but there
 is a long delay before the audio starts, basically for the
 first 6-10 seconds of the call there is dead silence,
 eventually the audio will start and everything works
 correctly. We never had this problem with the PRI. So I suspect
 it has something to do with a call coming in as SIP and going
 out as SIP.

 At first I thought it was a call forwarding issue because I got
 this message in the console: [Feb 26 12:35:19]
 NOTICE[1143][C-025d]: app_dial.c:958 do_forward: Not
 accepting call completion offers from call-forward recipient 
 Local/1XX@default-0013;1

 So I put this in my dial plan:

 1AA => { NoOp(${CALLERID(num)}); Ringing; 
 Set(CHANNEL(musicclass)=none); 
 Dial(${OUTBOUND-TRUNKR}/1XX,30); Voicemail(198,u); };

 So basically as soon as someone calls incoming number
 AA, Asterisk dials phone number XX. it's a
 quick and dirty way to call forward.. and this does the same
 thing, there's a good 8 second delay before the audio kicks
 in.

 There is a Linux firewall with NAT in the path, but I have no
 other audio issues, so don't *think* it's a factor. I just
 upgraded to asterisk 11.2.1.

 Asterisk 11.2.1 built by root @ phonesys2 on a i686 running
 Linux on 2013-02-23 01:40:02 UTC

 Any help would be appreciated, Thanks,

>>> 

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Re: [asterisk-users] Asterisk 1.8 and dual stack support

2013-03-21 Thread Michael L. Young

- Original Message -
> From: "Jaap Winius" 
> To: asterisk-users@lists.digium.com
> Sent: Thursday, March 21, 2013 12:47:57 PM
> Subject: [asterisk-users] Asterisk 1.8 and dual stack support
> 
> Hi folks,
> 
> Following an upgrade to Debian wheezy, I'm now running Asterisk
> 1.8.13.1.
> As opposed to Asterisk 1.6.2.9 that I ran with squeeze, this version
> can
> support IPv6. However, it seems that I can't get it to support both
> IPv4
> and IPv6 at the same time. For example, if in sip.conf I set the
> bindaddr
> variable to '::' it will only listen on IPv6 and none of my IPv4-only
> friends and peers will be able to connect to it. On the other hand,
> if I
> set it to '0.0.0.0' then it will not listen on IPv6.

How are you determining that it is not listening on IPv4?

bindaddr=:: should allow you to support dual stack.

Michael


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Re: [asterisk-users] Howto create variable from the name of another one and get content of it

2013-03-21 Thread Steve Edwards


On Thu, 21 Mar 2013, Administrator TOOTAI wrote:


I have a variable created like

... Set(__myVar=${ARG1})
... Set(__${myVar}STATUS=)

If ARG1 is abcd, variable is abcdSTATUS and should be empty. This is OK.

Now I would like to get the value of abcdSTATUS. How to do it? 
${${myVar}STATUS}} isn't working, nor ${{myvar}STATUS}


If 'variable is abcdSTATUS and should be empty' what value are you trying 
to get?


In your first try '${${myVar}STATUS}}' you have 1 too many closing brace. 
In your second try '${{myvar}STATUS}' you're missing a 'dollar.'


Is this what you are trying to do?

exten = *,n,set(ARG1=abcd)
exten = *,n,set(__myVar=${ARG1})
exten = *,n,set(__${myVar}STATUS=status)
exten = *,n,verbose(myVar = ${myVar})
exten = *,n,verbose(dollar{myVar}STATUS = ${${myVar}STATUS})
exten = *,n,hangup()

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Asterisk 1.8 and dual stack support

2013-03-21 Thread Rob van der Putten


Hi there


Michael L. Young wrote:


How are you determining that it is not listening on IPv4?

bindaddr=:: should allow you to support dual stack.


Which is the way it works with 1:1.8.11.1-1digium1~squeeze.
I even use Asterisk as a RTP audio IPv4 <-> IPv6 proxy.

Regards,
Rob


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Re: [asterisk-users] Allow/Disallow

2013-03-21 Thread Nick Khamis

Hello Asghar,

I fixed the issue after I realized that I was specifying allow before
disallow. Sorry for the noise!!!


Nick.

On 3/21/13, Asghar Mohammad  wrote:
> please post sip.conf.
>
> On Thu, Mar 21, 2013 at 8:01 PM, Nick Khamis  wrote:
>
>> Hello Everyone,
>>
>> I have disallow=all and allow=g729 set in sip.conf however, it seems
>> that asterisk still thinks it support other codecs:
>>
>> Capabilities: us - 0x8008000e (gsm|ulaw|alaw|h263|testlaw). How
>> can I disable gsm,ulaw,alaw.
>>
>> Thanks in Advance,
>>
>> Nick.
>>
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>

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Re: [asterisk-users] Delay before audio starts

2013-03-21 Thread Gerard

On 03/21/13 14:14, Gerard wrote:
>> I think a simple tcpdump of the traffic will show the mystery. It can
>> be your provider doing something nasty. Have you tried using some
>> other cheap SIP termination? or arrange a fake termination yourself
>> on another server?
>>
>> Leandro
> 
> I thought so too, but it doesn't appear to .
> 
> I just bought a door intercom device, set up the extension for it and
> it's doing the same thing, when it connects there is a 10 second delay
> before the other side can hear my voice.
> However watching tcpdump, the audio starts streaming both ways immediately.
> Changing the dialplan fixes the issue:
> 957 => { // Test door phone
> Answer(); //  <--- this line fixes the problem!
> Dial(SIP/199,20);
> Hangup();
> };
> 
> It's an ok workaround for the door intercom, but in the case of the
> forwarded calls below, as soon as I Answer() their ringback disappears
> and the line goes dead while they wait for our guy to answer the phone.
> 
> I may start a separate post about getting ringback to work after Answer();

As a followup, hold music instead of ringback works fine, so as my
current workaround, I'm using an mp3 of the ringback sound as the hold
music.
Anything is better then a dead line :)


> 
> Thanks for the help by the way.
> -Gerard
> 
> 
> On 03/01/13 14:34, Leandro Dardini wrote:
> 
>>
>> 2013/3/1 Gerard 
>>
>>> I thought it was the re-invites too, but I have it turned off
>>> everywhere.
>>>
>>> On 03/01/13 08:36, Eric Wieling wrote:
 When Answer fixes the issue, the root cause is often NAT (could
 be
>>> firewall) since Answering the call prevents any reinvites.

 -Original Message- From:
 asterisk-users-boun...@lists.digium.com [mailto:
>>> asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
 Sent: Friday, March 01, 2013 9:33 AM To:
 asterisk-users@lists.digium.com Subject: Re: [asterisk-users]
 Delay before audio starts

 I've found a workaround of sorts, If I change my below code to : 
 1AA => { NoOp(${CALLERID(num)}); Answer();  //
 <--- add this Ringing; 
 Set(CHANNEL(musicclass)=none); 
 Dial(${OUTBOUND-TRUNKR}/1XX,30); Voicemail(198,u); };

 That fixes the issue. It doesn't fix the call forward issue on
 the phone
>>> though. I've made a few extra extensions, one each corresponding to
>>> a number he wants to call forward to, if I have him forward to the
>>> extensions who then forward to the real number, it works, thanks to
>>> adding "Answer()" to the dialplan.

 -Gerard


 On 02/26/13 13:19, Gerard wrote:
> Hi everyone,
>
> I'm having a hard time figuring this issue out, we just
> switched from a T1 PRI to a SIP trunk provider and that's when
> the issue started. Now when someone forwards all calls on their
> phone to a cellphone, when a customer calls in, Asterisk
> correctly calls the cellphone and connects the call, but there
> is a long delay before the audio starts, basically for the
> first 6-10 seconds of the call there is dead silence,
> eventually the audio will start and everything works
> correctly. We never had this problem with the PRI. So I suspect
> it has something to do with a call coming in as SIP and going
> out as SIP.
>
> At first I thought it was a call forwarding issue because I got
> this message in the console: [Feb 26 12:35:19]
> NOTICE[1143][C-025d]: app_dial.c:958 do_forward: Not
> accepting call completion offers from call-forward recipient 
> Local/1XX@default-0013;1
>
> So I put this in my dial plan:
>
> 1AA => { NoOp(${CALLERID(num)}); Ringing; 
> Set(CHANNEL(musicclass)=none); 
> Dial(${OUTBOUND-TRUNKR}/1XX,30); Voicemail(198,u); };
>
> So basically as soon as someone calls incoming number
> AA, Asterisk dials phone number XX. it's a
> quick and dirty way to call forward.. and this does the same
> thing, there's a good 8 second delay before the audio kicks
> in.
>
>
> There is a Linux firewall with NAT in the path, but I have no
> other audio issues, so don't *think* it's a factor. I just
> upgraded to asterisk 11.2.1.
>
>
> Asterisk 11.2.1 built by root @ phonesys2 on a i686 running
> Linux on 2013-02-23 01:40:02 UTC
>
>
> Any help would be appreciated, Thanks,
>

> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
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> 


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Network

Re: [asterisk-users] Asterisk 1.8 and dual stack support

2013-03-21 Thread Jaap Winius

On Thu, 21 Mar 2013 12:21:44 -0700, Michael L. Young wrote:

> How are you determining that it is not listening on IPv4?
> 
> bindaddr=:: should allow you to support dual stack.

That's what I thought would happen. When I set bindaddr=:: and use 
'netstat -lpn |grep 5060' it shows:

  udp6 0   0 :::5060   :::* 9898/asterisk

Services like this usually also support IPv4 and as much is suggested by 
this comment in the sip.conf that comes with my Asterisk package:

  ; (Note that using bindaddr=:: will show only a single
  ; IPv6 socket in netstat. IPv4 is supported at the same
  ; time using IPv4-mapped IPv6 addresses.)

However, the moment I reload my sip.conf with bindaddr=::, my entire list 
of IPv4-only peers loses contact with Asterisk with warnings about the 
network being unreachable. So, it would appear that the version of 
Asterisk that I'm using is operating with a single stack socket.

Cheers,

Jaap

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Re: [asterisk-users] Howto create variable from the name of another one and get content of it

2013-03-21 Thread Administrator TOOTAI


Le 21/03/2013 20:27, Steve Edwards a écrit :

On Thu, 21 Mar 2013, Administrator TOOTAI wrote:


I have a variable created like

... Set(__myVar=${ARG1})
... Set(__${myVar}STATUS=)

If ARG1 is abcd, variable is abcdSTATUS and should be empty. This is OK.

Now I would like to get the value of abcdSTATUS. How to do it? 
${${myVar}STATUS}} isn't working, nor ${{myvar}STATUS}


If 'variable is abcdSTATUS and should be empty' what value are you 
trying to get?


In your first try '${${myVar}STATUS}}' you have 1 too many closing 
brace. In your second try '${{myvar}STATUS}' you're missing a 'dollar.'


Is this what you are trying to do?

exten = *,n,set(ARG1=abcd)
exten = *,n,set(__myVar=${ARG1})
exten = *,n,set(__${myVar}STATUS=status)
exten = *,n,verbose(myVar = ${myVar})
exten = *,n,verbose(dollar{myVar}STATUS = ${${myVar}STATUS})
exten = *,n,hangup()


Exactly, and this is working.

I found the problem, it's (I think) a bug with queue command. My dialplan:

[context]
...
exten => 33123,n,macro(unpauseQueueMembers,q820,104,105,136,,)
exten => 33123,n(back2Queue),Queue(${myQueue},nit,,,14400)
exten => 33123,n,NoOp(Queue ${myQueue} call status is ${QUEUESTATUS} - 
Dial status is ${DIALSTATUS} - Our status is ${${myQueue}STATUS})  ; 
value is empty

exten => 33123,n,GotoIf($["${${myQueue}STATUS}" = "myTIMEOUT"]?back2Queue)

[to-q820]

exten => 104,1,Dial(SIP/${EXTEN},,Tt)
exten => 105,1,Dial(SIP/${EXTEN},,Tt)
exten => 136,1,Dial(SIP/${EXTEN},,Tt)
exten => _XXX,2,macro(queueCallStatus,${EXTEN})

[macro-queueCallStatus]

exten => s,1,Set(__myExten=${ARG1})
   same => n,NoOp(Call status to ${myExten}@${myQueue} is ${DIALSTATUS})
   same => n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?exitMacro)
   same => n,GotoIf($["${DIALSTATUS}" != "NOANSWER"]?Pause)
   same => n(exitMacro),MacroExit
...
   same => 
n(Pause),NoOp(PauseQueueMember(,Local/${myExten}@to-${myQueue}))

   same => n,Set(__${myQueue}STATUS=myTIMEOUT)
   same => n,NoOp(Value of my variable is ${${myQueue}STATUS})  
 ; here I get correct value

...
[macro-unpauseQueueMembers]

exten => s,1,Set(__myQueue=${ARG1})
   same => n,Set(__${myQueue}STATUS=)
   same => n,UnpauseQueueMember(,Local/${ARG2}@to-${myQueue})
   same => n,UnpauseQueueMember(,Local/${ARG3}@to-${myQueue})
   same => n,UnpauseQueueMember(,Local/${ARG4}@to-${myQueue})

If I 33123  the NoOp(Value of my variable is ${${myQueue}STATUS}) in 
macro-queueCallStatus shows the right value which is text myTIMEOUT. But 
this value isn't anymore present in
NoOp(Queue ${myQueue} call status is ${QUEUESTATUS} - Dial status is 
${DIALSTATUS} - Our status is ${${myQueue}STATUS}) from [context] which 
is *after* the queue cmd exit.


This is Asterisk 10.11.1. Can someone confirm before opening a bug?

--
Daniel

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Re: [asterisk-users] Asterisk 1.8 and dual stack support

2013-03-21 Thread Asghar Mohammad

:)

On Thu, Mar 21, 2013 at 10:27 PM, Jaap Winius  wrote:

> On Thu, 21 Mar 2013 12:21:44 -0700, Michael L. Young wrote:
>
> > How are you determining that it is not listening on IPv4?
> >
> > bindaddr=:: should allow you to support dual stack.
>
> That's what I thought would happen. When I set bindaddr=:: and use
> 'netstat -lpn |grep 5060' it shows:
>
>   udp6 0   0 :::5060   :::* 9898/asterisk
>
> Services like this usually also support IPv4 and as much is suggested by
> this comment in the sip.conf that comes with my Asterisk package:
>
>   ; (Note that using bindaddr=:: will show only a single
>   ; IPv6 socket in netstat. IPv4 is supported at the same
>   ; time using IPv4-mapped IPv6 addresses.)
>
> However, the moment I reload my sip.conf with bindaddr=::, my entire list
> of IPv4-only peers loses contact with Asterisk with warnings about the
> network being unreachable. So, it would appear that the version of
> Asterisk that I'm using is operating with a single stack socket.
>
> Cheers,
>
> Jaap
>
>
> --
> _
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Re: [asterisk-users] Delay before audio starts

2013-03-21 Thread Asghar Mohammad

hi,
exten 000,1.Progress() work in some situation.

On Thu, Mar 21, 2013 at 9:30 PM, Gerard  wrote:

> On 03/21/13 14:14, Gerard wrote:
> >> I think a simple tcpdump of the traffic will show the mystery. It can
> >> be your provider doing something nasty. Have you tried using some
> >> other cheap SIP termination? or arrange a fake termination yourself
> >> on another server?
> >>
> >> Leandro
> >
> > I thought so too, but it doesn't appear to .
> >
> > I just bought a door intercom device, set up the extension for it and
> > it's doing the same thing, when it connects there is a 10 second delay
> > before the other side can hear my voice.
> > However watching tcpdump, the audio starts streaming both ways
> immediately.
> > Changing the dialplan fixes the issue:
> > 957 => { // Test door phone
> > Answer(); //  <--- this line fixes the problem!
> > Dial(SIP/199,20);
> > Hangup();
> > };
> >
> > It's an ok workaround for the door intercom, but in the case of the
> > forwarded calls below, as soon as I Answer() their ringback disappears
> > and the line goes dead while they wait for our guy to answer the phone.
> >
> > I may start a separate post about getting ringback to work after
> Answer();
>
> As a followup, hold music instead of ringback works fine, so as my
> current workaround, I'm using an mp3 of the ringback sound as the hold
> music.
> Anything is better then a dead line :)
>
>
> >
> > Thanks for the help by the way.
> > -Gerard
> >
> >
> > On 03/01/13 14:34, Leandro Dardini wrote:
> >
> >>
> >> 2013/3/1 Gerard 
> >>
> >>> I thought it was the re-invites too, but I have it turned off
> >>> everywhere.
> >>>
> >>> On 03/01/13 08:36, Eric Wieling wrote:
>  When Answer fixes the issue, the root cause is often NAT (could
>  be
> >>> firewall) since Answering the call prevents any reinvites.
> 
>  -Original Message- From:
>  asterisk-users-boun...@lists.digium.com [mailto:
> >>> asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
>  Sent: Friday, March 01, 2013 9:33 AM To:
>  asterisk-users@lists.digium.com Subject: Re: [asterisk-users]
>  Delay before audio starts
> 
>  I've found a workaround of sorts, If I change my below code to :
>  1AA => { NoOp(${CALLERID(num)}); Answer();  //
>  <--- add this Ringing;
>  Set(CHANNEL(musicclass)=none);
>  Dial(${OUTBOUND-TRUNKR}/1XX,30); Voicemail(198,u); };
> 
>  That fixes the issue. It doesn't fix the call forward issue on
>  the phone
> >>> though. I've made a few extra extensions, one each corresponding to
> >>> a number he wants to call forward to, if I have him forward to the
> >>> extensions who then forward to the real number, it works, thanks to
> >>> adding "Answer()" to the dialplan.
> 
>  -Gerard
> 
> 
>  On 02/26/13 13:19, Gerard wrote:
> > Hi everyone,
> >
> > I'm having a hard time figuring this issue out, we just
> > switched from a T1 PRI to a SIP trunk provider and that's when
> > the issue started. Now when someone forwards all calls on their
> > phone to a cellphone, when a customer calls in, Asterisk
> > correctly calls the cellphone and connects the call, but there
> > is a long delay before the audio starts, basically for the
> > first 6-10 seconds of the call there is dead silence,
> > eventually the audio will start and everything works
> > correctly. We never had this problem with the PRI. So I suspect
> > it has something to do with a call coming in as SIP and going
> > out as SIP.
> >
> > At first I thought it was a call forwarding issue because I got
> > this message in the console: [Feb 26 12:35:19]
> > NOTICE[1143][C-025d]: app_dial.c:958 do_forward: Not
> > accepting call completion offers from call-forward recipient
> > Local/1XX@default-0013;1
> >
> > So I put this in my dial plan:
> >
> > 1AA => { NoOp(${CALLERID(num)}); Ringing;
> > Set(CHANNEL(musicclass)=none);
> > Dial(${OUTBOUND-TRUNKR}/1XX,30); Voicemail(198,u); };
> >
> > So basically as soon as someone calls incoming number
> > AA, Asterisk dials phone number XX. it's a
> > quick and dirty way to call forward.. and this does the same
> > thing, there's a good 8 second delay before the audio kicks
> > in.
> >
> >
> > There is a Linux firewall with NAT in the path, but I have no
> > other audio issues, so don't *think* it's a factor. I just
> > upgraded to asterisk 11.2.1.
> >
> >
> > Asterisk 11.2.1 built by root @ phonesys2 on a i686 running
> > Linux on 2013-02-23 01:40:02 UTC
> >
> >
> > Any help would be appreciated, Thanks,
> >
> 
> >
> > --
> > _
> > -- Bandwidth and Coloc

Re: [asterisk-users] Howto create variable from the name of another one and get content of it

2013-03-21 Thread Richard Mudgett

> > On Thu, 21 Mar 2013, Administrator TOOTAI wrote:
> >
> >> I have a variable created like
> >>
> >> ... Set(__myVar=${ARG1})
> >> ... Set(__${myVar}STATUS=)
> >>
> >> If ARG1 is abcd, variable is abcdSTATUS and should be empty. This
> >> is OK.
> >>
> >> Now I would like to get the value of abcdSTATUS. How to do it?
> >> ${${myVar}STATUS}} isn't working, nor ${{myvar}STATUS}
> >
> > If 'variable is abcdSTATUS and should be empty' what value are you
> > trying to get?
> >
> > In your first try '${${myVar}STATUS}}' you have 1 too many closing
> > brace. In your second try '${{myvar}STATUS}' you're missing a
> > 'dollar.'
> >
> > Is this what you are trying to do?
> >
> > exten = *,n,set(ARG1=abcd)
> > exten = *,n,set(__myVar=${ARG1})
> > exten = *,n,set(__${myVar}STATUS=status)
> > exten = *,n,verbose(myVar = ${myVar})
> > exten = *,n,verbose(dollar{myVar}STATUS =
> > ${${myVar}STATUS})
> > exten = *,n,hangup()
> 
> Exactly, and this is working.
> 
> I found the problem, it's (I think) a bug with queue command. My
> dialplan:
> 
> [context]
> ...
> exten => 33123,n,macro(unpauseQueueMembers,q820,104,105,136,,)
> exten => 33123,n(back2Queue),Queue(${myQueue},nit,,,14400)
> exten => 33123,n,NoOp(Queue ${myQueue} call status is ${QUEUESTATUS}
> -
> Dial status is ${DIALSTATUS} - Our status is ${${myQueue}STATUS})  ;
> value is empty
> exten => 33123,n,GotoIf($["${${myQueue}STATUS}" =
> "myTIMEOUT"]?back2Queue)
> 
> [to-q820]
> 
> exten => 104,1,Dial(SIP/${EXTEN},,Tt)
> exten => 105,1,Dial(SIP/${EXTEN},,Tt)
> exten => 136,1,Dial(SIP/${EXTEN},,Tt)
> exten => _XXX,2,macro(queueCallStatus,${EXTEN})
> 
> [macro-queueCallStatus]
> 
> exten => s,1,Set(__myExten=${ARG1})
> same => n,NoOp(Call status to ${myExten}@${myQueue} is
> ${DIALSTATUS})
> same => n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?exitMacro)
> same => n,GotoIf($["${DIALSTATUS}" != "NOANSWER"]?Pause)
> same => n(exitMacro),MacroExit
> ...
> same =>
> n(Pause),NoOp(PauseQueueMember(,Local/${myExten}@to-${myQueue}))
> same => n,Set(__${myQueue}STATUS=myTIMEOUT)
> same => n,NoOp(Value of my variable is ${${myQueue}STATUS})
>   ; here I get correct value
> ...
> [macro-unpauseQueueMembers]
> 
> exten => s,1,Set(__myQueue=${ARG1})
> same => n,Set(__${myQueue}STATUS=)
> same => n,UnpauseQueueMember(,Local/${ARG2}@to-${myQueue})
> same => n,UnpauseQueueMember(,Local/${ARG3}@to-${myQueue})
> same => n,UnpauseQueueMember(,Local/${ARG4}@to-${myQueue})
> 
> If I 33123  the NoOp(Value of my variable is ${${myQueue}STATUS}) in
> macro-queueCallStatus shows the right value which is text myTIMEOUT.
> But
> this value isn't anymore present in
> NoOp(Queue ${myQueue} call status is ${QUEUESTATUS} - Dial status is
> ${DIALSTATUS} - Our status is ${${myQueue}STATUS}) from [context]
> which
> is *after* the queue cmd exit.

The the parent channel (the incoming channel on exten 33123) sets the
variable to empty.  A child channel (Dialed by app Queue) running a
local channel in the context [to-q820] sets the variable to the status
you are wanting.  The child cannot send that variable value back to the
parent channel.  Channel variable inheritance only goes one way: from
parent to child.

Richard

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Re: [asterisk-users] Asterisk 1.8 and dual stack support

2013-03-21 Thread Michael L. Young

- Original Message -
> From: "Jaap Winius" 
> To: asterisk-users@lists.digium.com
> Sent: Thursday, March 21, 2013 5:27:37 PM
> Subject: Re: [asterisk-users] Asterisk 1.8 and dual stack support
> 
> That's what I thought would happen. When I set bindaddr=:: and use
> 'netstat -lpn |grep 5060' it shows:
> 
>   udp6 0   0 :::5060   :::* 9898/asterisk
> 
> Services like this usually also support IPv4 and as much is suggested
> by
> this comment in the sip.conf that comes with my Asterisk package:
> 
>   ; (Note that using bindaddr=:: will show only a single
>   ; IPv6 socket in netstat. IPv4 is supported at the same
>   ; time using IPv4-mapped IPv6 addresses.)
> 
> However, the moment I reload my sip.conf with bindaddr=::, my entire
> list
> of IPv4-only peers loses contact with Asterisk with warnings about
> the
> network being unreachable. So, it would appear that the version of
> Asterisk that I'm using is operating with a single stack socket.

Let me try to understand this.  With bindaddr set as "bindaddr=::", upon 
starting Asterisk, you are fine and all your IPv4 peers connect properly.  
Therefore, dual stack is working at this point.  Upon issuing a "sip reload", 
your peers lose their ability to communicate with Asterisk?  Is that correct?  
What does "netstat -lpn |grep 5060" show after the reload?

These "network unreachable" warnings are from Asterisk or your peers?

What version of Asterisk are you using?

Asterisk 1.8.0 had IPv6 support in it.  Therefore, every minor version released 
since would still have IPv6 support in it.

Michael

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Re: [asterisk-users] Asterisk 1.8 and dual stack support

2013-03-21 Thread Jaap Winius

On Thu, 21 Mar 2013 16:02:17 -0700, Michael L. Young wrote:

> Let me try to understand this.  With bindaddr set as "bindaddr=::", upon
> starting Asterisk, you are fine and all your IPv4 peers connect
> properly. Therefore, dual stack is working at this point. ...

You minunderstand. When I start Asterisk with "bindaddr=::", the netstat 
output shows that it's using udp6, which usually means that the service 
is running in dual stack mode, but this is apparently not the case. On my 
system, it really is only listening on IPv6. That's why I said that, 
despite appearances, as soon as I reload SIP (or restart Asterisk) with 
this setting, I lose contact with my entire list of IPv4-only peers, 
while Asterisk gives warnings about the network being unreachable (the 
IPv4 network).

I've also tried using multiple bindaddr lines with a mix of IPv4 and IPv6 
addresses, but then the service ends up binding only to the last address. 
Therefore, it looks to me like the version of Asterisk that I'm running 
is only capable of running in single stack mode, supporting either IPv4 
or IPv6, but not both at the same time.

> Upon issuing a "sip reload", your peers lose their ability
> to communicate with Asterisk? Is that correct?

That's right.

> What does "netstat -lpn |grep 5060" show after the reload?

  udp6 0   0 :::5060   :::* 9898/asterisk

> These "network unreachable" warnings are from Asterisk or your peers?

>From Asterisk. They look like this for two of my IPv4 SIP devices:

[Mar 21 23:24:18] NOTICE[9931]: chan_sip.c:26242 sip_poke_noanswer: Peer 
'1000' is now UNREACHABLE!  Last qualify: 110
[Mar 21 23:24:18] NOTICE[9931]: chan_sip.c:26242 sip_poke_noanswer: Peer 
'patton' is now UNREACHABLE!  Last qualify: 20

I also get errors for connections to SIP servers for which I have 
"register" entries in the [general] section of sip.conf. The errors for 
one of them, sip.xs4all.nl, which is IPv4 only, look like this:

[Mar 21 23:24:14] ERROR[9931]: netsock2.c:263 ast_sockaddr_resolve: 
getaddrinfo("sip.xs4all.nl", "(null)", ...): No address associated with 
hostname
[Mar 21 23:24:14] WARNING[9931]: acl.c:582 resolve_first: Unable to 
lookup 'sip.xs4all.nl'

Anyway, as soon as I reload sip without "bindaddr=::", these errors stop.

> What version of Asterisk are you using?

Version 1.8.13.1.

> Asterisk 1.8.0 had IPv6 support in it.  Therefore, every minor version
> released since would still have IPv6 support in it.

That's good to know, so maybe it's just my minor version that has a bug 
that prevents it from running in dual stack mode. That's what my question 
was about in the first place.

Cheers,

Jaap

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Re: [asterisk-users] SIP account registration fails after upgrade to 1.8

2013-03-21 Thread Jaap Winius

On Thu, 21 Mar 2013 16:35:16 +, Jaap Winius wrote:

> Hopefully, my ISP will see fit to squash this bug ASAP.

Well, I got my answer from them quickly enough: Nope.

Luckily, somebody was kind enough to suggest a workaround. Unfortunately, 
it involves, downloading the source code and making a few changes to it 
to prevent Asterisk from adding '@' to the end of the Call-ID 
string. Nevertheless, it's easy enough to do. The idea is to look for 
this string that appears twice in ./channels/chan_sip.c:

  ast_string_field_build(pvt, callid, "%s@%s",
  generate_random_string(buf, sizeof(buf)), host);

And to change it to:

  ast_string_field_build(pvt, callid, "%s",
  generate_random_string(buf, sizeof(buf)));

Now my Call-IDs look like this:

   Call-ID: 63935a8d2144d4f1309024fd7612f608

Instead of this:

   Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:abcd:1::a]

Still, I'd much prefer that my ISP fixed the problem instead, because now 
every time a security update becomes available for Asterisk, I'm going to 
have to download the source code, make the same changes, recompile it and 
install it all over again and again. Ho hum.

Of course, an even better solution would be if Asterisk had a variable 
with which to alter the Call-ID string format so that I could omit the IP 
address. :-)

Cheers,

Jaap


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Re: [asterisk-users] Howto create variable from the name of another one and get content of it

2013-03-21 Thread Satish Barot

>
>
> > I found the problem, it's (I think) a bug with queue command. My
> > dialplan:
> >
> > [context]
> > ...
> > exten => 33123,n,macro(unpauseQueueMembers,q820,104,105,136,,)
> > exten => 33123,n(back2Queue),Queue(${myQueue},nit,,,14400)
> > exten => 33123,n,NoOp(Queue ${myQueue} call status is ${QUEUESTATUS}
> > -
> > Dial status is ${DIALSTATUS} - Our status is ${${myQueue}STATUS})  ;
> > value is empty
> > exten => 33123,n,GotoIf($["${${myQueue}STATUS}" =
> > "myTIMEOUT"]?back2Queue)
> >
> > [to-q820]
> >
> > exten => 104,1,Dial(SIP/${EXTEN},,Tt)
> > exten => 105,1,Dial(SIP/${EXTEN},,Tt)
> > exten => 136,1,Dial(SIP/${EXTEN},,Tt)
> > exten => _XXX,2,macro(queueCallStatus,${EXTEN})
> >
> > [macro-queueCallStatus]
> >
> > exten => s,1,Set(__myExten=${ARG1})
> > same => n,NoOp(Call status to ${myExten}@${myQueue} is
> > ${DIALSTATUS})
> > same => n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?exitMacro)
> > same => n,GotoIf($["${DIALSTATUS}" != "NOANSWER"]?Pause)
> > same => n(exitMacro),MacroExit
> > ...
> > same =>
> > n(Pause),NoOp(PauseQueueMember(,Local/${myExten}@to-${myQueue}))
> > same => n,Set(__${myQueue}STATUS=myTIMEOUT)
> > same => n,NoOp(Value of my variable is ${${myQueue}STATUS})
> >   ; here I get correct value
> > ...
> > [macro-unpauseQueueMembers]
> >
> > exten => s,1,Set(__myQueue=${ARG1})
> > same => n,Set(__${myQueue}STATUS=)
> > same => n,UnpauseQueueMember(,Local/${ARG2}@to-${myQueue})
> > same => n,UnpauseQueueMember(,Local/${ARG3}@to-${myQueue})
> > same => n,UnpauseQueueMember(,Local/${ARG4}@to-${myQueue})
> >
> > If I 33123  the NoOp(Value of my variable is ${${myQueue}STATUS}) in
> > macro-queueCallStatus shows the right value which is text myTIMEOUT.
> > But
> > this value isn't anymore present in
> > NoOp(Queue ${myQueue} call status is ${QUEUESTATUS} - Dial status is
> > ${DIALSTATUS} - Our status is ${${myQueue}STATUS}) from [context]
> > which
> > is *after* the queue cmd exit.
>
> The the parent channel (the incoming channel on exten 33123) sets the
> variable to empty.  A child channel (Dialed by app Queue) running a
> local channel in the context [to-q820] sets the variable to the status
> you are wanting.  The child cannot send that variable value back to the
> parent channel.  Channel variable inheritance only goes one way: from
> parent to child.
>
> Richard
>
>
SHARED function is your best pal here.

Here is the updated (but not tested) dialplan for you

[context]
...
;Store CHANNEL in a variable
exten => 33123,n,Set(__PARENTCHANNEL=${CHANNEL})
exten => 33123,n,macro(unpauseQueueMembers,q820,104,105,136,,)
exten => 33123,n(back2Queue),Queue(${myQueue},nit,,,14400)
exten => 33123,n,NoOp(Queue ${myQueue} call status is ${QUEUESTATUS} - Dial
status is ${DIALSTATUS} - Our status is ${${myQueue}STATUS}) ; value is
empty
;;exten => 33123,n,GotoIf($["${${myQueue}STATUS}" = "myTIMEOUT"]?back2Queue)
; See the SHARED FUNCTION
exten => 33123,n,GotoIf($["${SHARED(${myQueue}STATUS,${PARENTCHANNEL})}" =
"myTIMEOUT"]?back2Queue)

[to-q820]

exten => 104,1,Dial(SIP/${EXTEN},,Tt)
exten => 105,1,Dial(SIP/${EXTEN},,Tt)
exten => 136,1,Dial(SIP/${EXTEN},,Tt)
exten => _XXX,2,macro(queueCallStatus,${EXTEN})

[macro-queueCallStatus]

exten => s,1,Set(__myExten=${ARG1})
   same => n,NoOp(Call status to ${myExten}@${myQueue} is ${DIALSTATUS})
   same => n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?exitMacro)
   same => n,GotoIf($["${DIALSTATUS}" != "NOANSWER"]?Pause)
   same => n(exitMacro),MacroExit
...
same => n(Pause),NoOp(PauseQueueMember(,Local/${myExten}@to-${myQueue}))
;;same => n,Set(__${myQueue}STATUS=myTIMEOUT)
same => n,Set(SHARED(${myQueue}STATUS,${PARENTCHANNEL})=myTIMEOUT)
;;same => n,NoOp(Value of my variable is ${${myQueue}STATUS}) ; here I get
correct value
; ${myQueue}STATUS is set here through SHARED FUNCTION
same => n,NoOp(Value of my variable is
${SHARED(${myQueue}STATUS,${PARENTCHANNEL})})
...
[macro-unpauseQueueMembers]

exten => s,1,Set(__myQueue=${ARG1})
   same => n,Set(__${myQueue}STATUS=)
   same => n,UnpauseQueueMember(,Local/${ARG2}@to-${myQueue})
   same => n,UnpauseQueueMember(,Local/${ARG3}@to-${myQueue})
   same => n,UnpauseQueueMember(,Local/${ARG4}@to-${myQueue})

Hope this helps.

--Satish Barot
Ahmedabad, India
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