Re: [asterisk-users] PJSIP issues with handling incoming calls

[Nick Awesome]

Ok, thanks for an answer. That solution works.

On 02 Sep 2014, at 22:36, Rainer Piper  wrote:

> contact_user in pjsip.conf has to point to the filter or to an agi in the 
> extentions.conf
> like:
> 
> pjsip.conf
> contact_user=blablabla
> 
> extensions.conf
> exten => blablabla, 1,NoOp(*** ${PJSIP_HEADER(read,To)} *** ${CALLERID(num)} 
> ***)
> 
> 
> Am 02.09.2014 um 20:11 schrieb Rainer Piper:
>> contact_user can be anything and calling an agi is no problem 
>> 
>> 
>> Am 02.09.2014 um 19:49 schrieb Nick Awesome:
>>> Okay, contact_user seems like do the job. Thanks
>>> is contact_user can be anything, or it should be same as username ?
>>> I would like to use contact_user for transmitting trunk name into agi script
>>> 
>>> On Sep 2, 2014, at 7:04 PM, Rainer Piper  wrote:
>>> 
 I use in pjsip.conf 
 [sipgate1]
 type=registration
 transport=transport-udp
 outbound_auth=sipgate1_auth
 server_uri=sip:sipgate.de
 client_uri=sip:555123...@sipgate.de
 contact_user=sipgatefilter ; goto the filter in extensions.conf
 retry_interval=60
 forbidden_retry_interval=600
 expiration=3600
 
 extensions.conf ; i'm cutting the dialed number out of the invite Header 
 and goto/jump to the extensions
 ; incoming VOIP 9716716x SIPGATE
 exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} *** 
 ${CALLERID(num)} ***)
 same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
 same => n,NoOp( 49${gotoadr:-11} )
 same => n,Goto(49${gotoadr:-11},1)
 
 ; the filter is jumping to the extensions ...
 
 ; incoming VOIP 97167160 SIPGATE -> MENU
 exten => 
 4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r)
 ; incoming VOIP 97167161 SIPGATE
 exten => 
 4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r)
 ; incoming VOIP 97167162 SIPGATE ECHO TEST
 exten => 
 4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
 ; incoming VOIP 97167163 SIPGATE
 exten => 
 4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
 ; incoming VOIP 97167164 SIPGATE
 exten => 
 4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
 ; incoming VOIP 97167165 SIPGATE
 exten => 
 4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
 ; incncoming VOIP 97167166 Mailbox
 exten => 
 4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
 ; incoming VOIP 97167167 CONF. 1
 exten => 
 4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
 ; incoming VOIP 97167168 CONF. 2
 ;exten => 
 4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
 exten => 4922897167168,1,Answer
 same => n,echo()
 same => n,Hangup()
 ; incoming VOIP 97167169 FAX
 ;exten => 
 4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
 
 
 Regards
 Rainer
 
 Am 02.09.2014 um 15:08 schrieb Joshua Colp:
> Nick Awesome wrote: 
>> register =>  73432260005:pass@10001 
>> register =>  73432260050:pass@10002 
>> 
>> [10001] 
>> type=peer 
>> host=80.75.132.66 
>> context=dialmap 
>> [10002] 
>> type=peer 
>> host=80.75.132.66 
>> context=dialmap 
> 
> Can you provide a sip debug of calls to both of these? I'm confused how 
> that... works... 
> 
 
 
 -- 
 Rainer Piper 
 Integration engineer 
 Koeslinstr. 56 
 53123 BONN 
 GERMANY 
 Phone: +49 228 97167161 
 P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
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>>> 
>>> 
>>> 
>> 
>> 
>> -- 
>> Rainer Piper 
>> Integration engineer 
>> Koeslinstr. 56 
>> 53123 BONN 
>> GERMANY 
>> Phone: +49 228 97167161 
>> P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
>> 
>> 
> 
> 
> -- 
> Rainer Piper 
> Integration engineer 
> Koeslinstr. 56 
> 53123 BONN 
> GERMANY 
> Phone: +49 228 97167161 
> P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
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Re: [asterisk-users] Setup Own IP PBX Server

[Chandran Manikandan]

Hi Patrick,
Thanks for your help. Let me try your advise and come back to you if need
further any assistance.


On Tue, Sep 2, 2014 at 9:41 PM, Patrick Laimbock 
wrote:

> On 01-09-14 12:31, Chandran Manikandan wrote:
> [snip]
>
>  I have installed Freepbx server and tried to configure sip extension.
>> It's working fine.
>>
>
> A better place for FreePBX related questions and to get help is:
> http://community.freepbx.org/
> Or hire their professional FreePBX support:
> http://www.freepbx.org/support-and-professional-services
>
> If you want to learn more about Asterisk in general then a good start is
> to first read "Asterisk: The Definitive Guide, 4th Edition" and go through
> the wiki at http://wiki.asterisk.org.
>
> HTH,
> Patrick
>
>
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>



-- 
*Thanks,*
*Manikandan.C*
*System Administrator*
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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Eric Wieling]

"core show codecs" does not show VP8 on my Asterisk 11.  I don't recall why we 
are not using H.264.  The novelty wore off long ago and few of our staff use 
video calling anymore.


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Tuesday, September 02, 2014 9:36 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

On 02-09-14 22:52, Eric Wieling wrote:
> A co-worker was doing video, I dislike video.  The phones were Polycom VVX, 
> The settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP 
> Settings / Video section we have Video: Enabled, H.263 and H.263p are the 
> only two video codecs enabled.

Thanks Eric. The obvious difference is that your co-worker was using 
H.263/H.263p while Bria uses H.264 or VP8. videosupport=yes was present 
in my sip.conf so it might be the codec. Time for more tinkering.

Thanks,
Patrick

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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Patrick Laimbock]

On 02-09-14 22:52, Eric Wieling wrote:

A co-worker was doing video, I dislike video.  The phones were Polycom VVX, The 
settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP 
Settings / Video section we have Video: Enabled, H.263 and H.263p are the only 
two video codecs enabled.


Thanks Eric. The obvious difference is that your co-worker was using 
H.263/H.263p while Bria uses H.264 or VP8. videosupport=yes was present 
in my sip.conf so it might be the codec. Time for more tinkering.


Thanks,
Patrick

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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Jeff LaCoursiere]

Don't forget "videosupport=yes" in sip.conf.

j


On 09/02/2014 03:52 PM, Eric Wieling wrote:

A co-worker was doing video, I dislike video.  The phones were Polycom VVX, The 
settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP 
Settings / Video section we have Video: Enabled, H.263 and H.263p are the only 
two video codecs enabled.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Tuesday, September 02, 2014 7:54 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

On 02-09-14 21:15, Eric Wieling wrote:

As long as you are NOT transcoding video should work in Asterisk.

Both apps were configured with identical (codec) settings so I don't see
how it would require transcoding. If you did get it to work I would
appreciate it if you could tell me which clients you used, the Asterisk
version, the OS and the relevant Asterisk config.

Thanks,
Patrick



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Tuesday, September 02, 2014 6:39 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

On 02-09-14 20:18, Khalid Touati wrote:

so it seems Asterisk Versions does not support video I guess

On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the
Bria app on Android and iPhone. With SELinux and the firewall
temporarily disabled I couldn't get it to work with either H264 or VP8.

HTH,
Patrick






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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Shishir Pokharel]

You might want to check if "videosupport=yes" in sip.conf

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Tuesday, September 02, 2014 1:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

A co-worker was doing video, I dislike video.  The phones were Polycom VVX, The 
settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP 
Settings / Video section we have Video: Enabled, H.263 and H.263p are the only 
two video codecs enabled.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Tuesday, September 02, 2014 7:54 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

On 02-09-14 21:15, Eric Wieling wrote:
> As long as you are NOT transcoding video should work in Asterisk.

Both apps were configured with identical (codec) settings so I don't see how it 
would require transcoding. If you did get it to work I would appreciate it if 
you could tell me which clients you used, the Asterisk version, the OS and the 
relevant Asterisk config.

Thanks,
Patrick


> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick 
> Laimbock
> Sent: Tuesday, September 02, 2014 6:39 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls
>
> On 02-09-14 20:18, Khalid Touati wrote:
>> so it seems Asterisk Versions does not support video I guess
>
> On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the 
> Bria app on Android and iPhone. With SELinux and the firewall 
> temporarily disabled I couldn't get it to work with either H264 or VP8.
>
> HTH,
> Patrick
>


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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Eric Wieling]

A co-worker was doing video, I dislike video.  The phones were Polycom VVX, The 
settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP 
Settings / Video section we have Video: Enabled, H.263 and H.263p are the only 
two video codecs enabled.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Tuesday, September 02, 2014 7:54 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

On 02-09-14 21:15, Eric Wieling wrote:
> As long as you are NOT transcoding video should work in Asterisk.

Both apps were configured with identical (codec) settings so I don't see 
how it would require transcoding. If you did get it to work I would 
appreciate it if you could tell me which clients you used, the Asterisk 
version, the OS and the relevant Asterisk config.

Thanks,
Patrick


> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
> Sent: Tuesday, September 02, 2014 6:39 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls
>
> On 02-09-14 20:18, Khalid Touati wrote:
>> so it seems Asterisk Versions does not support video I guess
>
> On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the
> Bria app on Android and iPhone. With SELinux and the firewall
> temporarily disabled I couldn't get it to work with either H264 or VP8.
>
> HTH,
> Patrick
>


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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Jeff LaCoursiere]

On 09/02/2014 03:14 PM, Administrator TOOTAI wrote:

Le 02/09/2014 20:18, Khalid Touati a écrit :

so it seems Asterisk Versions does not support video I guess


Asterisk supports video. I'm using it with asterisk 1.4 1.8 and 11 
with GrandStream phones (H263, H263+ and H264). Works perfectly




I can second that with GS phones, asterisk 1.4 and 1.8.

j


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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Administrator TOOTAI]

Le 02/09/2014 20:18, Khalid Touati a écrit :

so it seems Asterisk Versions does not support video I guess


Asterisk supports video. I'm using it with asterisk 1.4 1.8 and 11 with 
GrandStream phones (H263, H263+ and H264). Works perfectly





On Mon, Sep 1, 2014 at 9:30 AM, Khalid Touati mailto:khalidtou...@gmail.com>> wrote:

Any article that goes through this (seems to be tedious) task to add
video support and patents?


On Mon, Sep 1, 2014 at 8:14 AM, Joshua Colp mailto:jc...@digium.com>> wrote:

Khalid Touati wrote:

Hi Guys,


Kia ora,


Do you know of any asterisk community version that does
video codec
trans-coding or in other words supports video? I have
1.8.8.1 and I see
h263.c format files but can't see that codec in make
menuselect. it
might be just a license issue (if h263 has to have license),
but not
sure if community versions offer video calls at all.


Video transcoding is both usually patent encumbered as well as
computationally expensive. Asterisk supports passing through the
video untouched, but that's about it.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com  &
www.asterisk.org 

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--
Khalid Touati




--
Khalid Touati





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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Patrick Laimbock]

On 02-09-14 21:15, Eric Wieling wrote:

As long as you are NOT transcoding video should work in Asterisk.


Both apps were configured with identical (codec) settings so I don't see 
how it would require transcoding. If you did get it to work I would 
appreciate it if you could tell me which clients you used, the Asterisk 
version, the OS and the relevant Asterisk config.


Thanks,
Patrick



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Tuesday, September 02, 2014 6:39 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

On 02-09-14 20:18, Khalid Touati wrote:

so it seems Asterisk Versions does not support video I guess


On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the
Bria app on Android and iPhone. With SELinux and the firewall
temporarily disabled I couldn't get it to work with either H264 or VP8.

HTH,
Patrick




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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Eric Wieling]

As long as you are NOT transcoding video should work in Asterisk.


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Tuesday, September 02, 2014 6:39 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

On 02-09-14 20:18, Khalid Touati wrote:
> so it seems Asterisk Versions does not support video I guess

On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the 
Bria app on Android and iPhone. With SELinux and the firewall 
temporarily disabled I couldn't get it to work with either H264 or VP8.

HTH,
Patrick

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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Patrick Laimbock]

On 02-09-14 20:18, Khalid Touati wrote:

so it seems Asterisk Versions does not support video I guess


On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the 
Bria app on Android and iPhone. With SELinux and the firewall 
temporarily disabled I couldn't get it to work with either H264 or VP8.


HTH,
Patrick

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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Rainer Piper]

contact_user in pjsip.conf has to point to the filter or to an agi in 
the extentions.conf

like:

pjsip.conf
contact_user=*blablabla

extensions.conf
**exten => blablabla, 1,NoOp(*** ${PJSIP_HEADER(read,To)} *** 
${CALLERID(num)} ***)

*

Am 02.09.2014 um 20:11 schrieb Rainer Piper:

contact_user can be anything and calling an agi is no problem


Am 02.09.2014 um 19:49 schrieb Nick Awesome:

Okay, contact_user seems like do the job. Thanks
is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi 
script


On Sep 2, 2014, at 7:04 PM, Rainer Piper > wrote:



I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600

*extensions.conf* ; i'm cutting the dialed number out of the invite 
Header and goto/jump to the extensions

; incoming VOIP 9716716x SIPGATE
exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} *** 
${CALLERID(num)} ***)

same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same => n,NoOp( 49${gotoadr:-11} )
same => n,*Goto(49${gotoadr:-11},1)*

; the filter is jumping to the extensions ...

; incoming VOIP 97167160 SIPGATE -> MENU
exten => 
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r)

; incoming VOIP 97167161 SIPGATE
exten => 
4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r)

; incoming VOIP 97167162 SIPGATE ECHO TEST
exten => 
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167163 SIPGATE
exten => 
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167164 SIPGATE
exten => 
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167165 SIPGATE
exten => 
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incncoming VOIP 97167166 Mailbox
exten => 
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167167 CONF. 1
exten => 
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167168 CONF. 2
;exten => 
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

exten => 4922897167168,1,Answer
same => n,echo()
same => n,Hangup()
; incoming VOIP 97167169 FAX
;exten => 
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)



Regards
Rainer

Am 02.09.2014 um 15:08 schrieb Joshua Colp:

Nick Awesome wrote:

register => 73432260005:pass@10001
register =>  73432260050:pass@10002

[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap


Can you provide a sip debug of calls to both of these? I'm confused 
how that... works...





--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
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--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)





--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
-- 
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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Rainer Piper]

Am 02.09.2014 um 20:11 schrieb Rainer Piper:

username ?



--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Khalid Touati]

so it seems Asterisk Versions does not support video I guess


On Mon, Sep 1, 2014 at 9:30 AM, Khalid Touati 
wrote:

> Any article that goes through this (seems to be tedious) task to add video
> support and patents?
>
>
> On Mon, Sep 1, 2014 at 8:14 AM, Joshua Colp  wrote:
>
>> Khalid Touati wrote:
>>
>>> Hi Guys,
>>>
>>
>> Kia ora,
>>
>>
>>  Do you know of any asterisk community version that does video codec
>>> trans-coding or in other words supports video? I have 1.8.8.1 and I see
>>> h263.c format files but can't see that codec in make menuselect. it
>>> might be just a license issue (if h263 has to have license), but not
>>> sure if community versions offer video calls at all.
>>>
>>
>> Video transcoding is both usually patent encumbered as well as
>> computationally expensive. Asterisk supports passing through the video
>> untouched, but that's about it.
>>
>> Cheers,
>>
>> --
>> Joshua Colp
>> Digium, Inc. | Senior Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.org
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> Khalid Touati
>
>


-- 
Khalid Touati
-- 
_
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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Rainer Piper]

contact_user can be anything and calling an agi is no problem


Am 02.09.2014 um 19:49 schrieb Nick Awesome:

Okay, contact_user seems like do the job. Thanks
is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi 
script


On Sep 2, 2014, at 7:04 PM, Rainer Piper > wrote:



I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600

*extensions.conf* ; i'm cutting the dialed number out of the invite 
Header and goto/jump to the extensions

; incoming VOIP 9716716x SIPGATE
exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} *** 
${CALLERID(num)} ***)

same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same => n,NoOp( 49${gotoadr:-11} )
same => n,*Goto(49${gotoadr:-11},1)*

; the filter is jumping to the extensions ...

; incoming VOIP 97167160 SIPGATE -> MENU
exten => 
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r)

; incoming VOIP 97167161 SIPGATE
exten => 
4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r)

; incoming VOIP 97167162 SIPGATE ECHO TEST
exten => 
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167163 SIPGATE
exten => 
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167164 SIPGATE
exten => 
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167165 SIPGATE
exten => 
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incncoming VOIP 97167166 Mailbox
exten => 
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167167 CONF. 1
exten => 
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167168 CONF. 2
;exten => 
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

exten => 4922897167168,1,Answer
same => n,echo()
same => n,Hangup()
; incoming VOIP 97167169 FAX
;exten => 
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)



Regards
Rainer

Am 02.09.2014 um 15:08 schrieb Joshua Colp:

Nick Awesome wrote:

register => 73432260005:pass@10001
register =>  73432260050:pass@10002

[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap


Can you provide a sip debug of calls to both of these? I'm confused 
how that... works...





--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
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http://lists.digium.com/mailman/listinfo/asterisk-users







--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
-- 
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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Nick Awesome]

Okay, contact_user seems like do the job. Thanks
is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi script

On Sep 2, 2014, at 7:04 PM, Rainer Piper  wrote:

> I use in pjsip.conf 
> [sipgate1]
> type=registration
> transport=transport-udp
> outbound_auth=sipgate1_auth
> server_uri=sip:sipgate.de
> client_uri=sip:555123...@sipgate.de
> contact_user=sipgatefilter ; goto the filter in extensions.conf
> retry_interval=60
> forbidden_retry_interval=600
> expiration=3600
> 
> extensions.conf ; i'm cutting the dialed number out of the invite Header and 
> goto/jump to the extensions
> ; incoming VOIP 9716716x SIPGATE
> exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} *** 
> ${CALLERID(num)} ***)
> same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
> same => n,NoOp( 49${gotoadr:-11} )
> same => n,Goto(49${gotoadr:-11},1)
> 
> ; the filter is jumping to the extensions ...
> 
> ; incoming VOIP 97167160 SIPGATE -> MENU
> exten => 
> 4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r)
> ; incoming VOIP 97167161 SIPGATE
> exten => 4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r)
> ; incoming VOIP 97167162 SIPGATE ECHO TEST
> exten => 
> 4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> ; incoming VOIP 97167163 SIPGATE
> exten => 
> 4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> ; incoming VOIP 97167164 SIPGATE
> exten => 
> 4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> ; incoming VOIP 97167165 SIPGATE
> exten => 
> 4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> ; incncoming VOIP 97167166 Mailbox
> exten => 
> 4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> ; incoming VOIP 97167167 CONF. 1
> exten => 
> 4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> ; incoming VOIP 97167168 CONF. 2
> ;exten => 
> 4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> exten => 4922897167168,1,Answer
> same => n,echo()
> same => n,Hangup()
> ; incoming VOIP 97167169 FAX
> ;exten => 4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> 
> 
> Regards
> Rainer
> 
> Am 02.09.2014 um 15:08 schrieb Joshua Colp:
>> Nick Awesome wrote: 
>>> register =>  73432260005:pass@10001 
>>> register =>  73432260050:pass@10002 
>>> 
>>> [10001] 
>>> type=peer 
>>> host=80.75.132.66 
>>> context=dialmap 
>>> [10002] 
>>> type=peer 
>>> host=80.75.132.66 
>>> context=dialmap 
>> 
>> Can you provide a sip debug of calls to both of these? I'm confused how 
>> that... works... 
>> 
> 
> 
> -- 
> Rainer Piper 
> Integration engineer 
> Koeslinstr. 56 
> 53123 BONN 
> GERMANY 
> Phone: +49 228 97167161 
> P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
> -- 
> _
> -- 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] PJSIP issues with handling incoming calls

[Rainer Piper]

upps  and delete the 49 in *Goto(49${gotoadr:-11},1) and *NoOp( 
49${gotoadr:-11} )

*just look at the cli output*

Am 02.09.2014 um 17:25 schrieb Rainer Piper:
PS all incoming calls are directed to sipgatefilter in extentions.conf 
and then filtered.
You maid have to adjust the -11 in *Goto(49${gotoadr:-11},1) ... just 
look at the cli output *NoOp( 49${gotoadr:-11} )


Am 02.09.2014 um 17:04 schrieb Rainer Piper:

I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600

*extensions.conf* ; i'm cutting the dialed number out of the invite 
Header and goto/jump to the extensions

; incoming VOIP 9716716x SIPGATE
exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} *** 
${CALLERID(num)} ***)

same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same => n,NoOp( 49${gotoadr:-11} )
same => n,*Goto(49${gotoadr:-11},1)*

; the filter is jumping to the extensions ...

; incoming VOIP 97167160 SIPGATE -> MENU
exten => 
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r)

; incoming VOIP 97167161 SIPGATE
exten => 
4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r)

; incoming VOIP 97167162 SIPGATE ECHO TEST
exten => 
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167163 SIPGATE
exten => 
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167164 SIPGATE
exten => 
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167165 SIPGATE
exten => 
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incncoming VOIP 97167166 Mailbox
exten => 
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167167 CONF. 1
exten => 
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167168 CONF. 2
;exten => 
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

exten => 4922897167168,1,Answer
same => n,echo()
same => n,Hangup()
; incoming VOIP 97167169 FAX
;exten => 
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)



Regards
Rainer

Am 02.09.2014 um 15:08 schrieb Joshua Colp:

Nick Awesome wrote:

register =>  73432260005:pass@10001
register =>  73432260050:pass@10002

[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap


Can you provide a sip debug of calls to both of these? I'm confused 
how that... works...





--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)





--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)





--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
-- 
_
-- 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

Re: [asterisk-users] PJSIP issues with handling incoming calls

[Rainer Piper]

PS all incoming calls are directed to sipgatefilter in extentions.conf 
and then filtered.
You maid have to adjust the -11 in *Goto(49${gotoadr:-11},1) ... just 
look at the cli output *NoOp( 49${gotoadr:-11} )


Am 02.09.2014 um 17:04 schrieb Rainer Piper:

I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600

*extensions.conf* ; i'm cutting the dialed number out of the invite 
Header and goto/jump to the extensions

; incoming VOIP 9716716x SIPGATE
exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} *** 
${CALLERID(num)} ***)

same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same => n,NoOp( 49${gotoadr:-11} )
same => n,*Goto(49${gotoadr:-11},1)*

; the filter is jumping to the extensions ...

; incoming VOIP 97167160 SIPGATE -> MENU
exten => 
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r)

; incoming VOIP 97167161 SIPGATE
exten => 
4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r)

; incoming VOIP 97167162 SIPGATE ECHO TEST
exten => 
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167163 SIPGATE
exten => 
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167164 SIPGATE
exten => 
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167165 SIPGATE
exten => 
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incncoming VOIP 97167166 Mailbox
exten => 
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167167 CONF. 1
exten => 
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167168 CONF. 2
;exten => 
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

exten => 4922897167168,1,Answer
same => n,echo()
same => n,Hangup()
; incoming VOIP 97167169 FAX
;exten => 
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)



Regards
Rainer

Am 02.09.2014 um 15:08 schrieb Joshua Colp:

Nick Awesome wrote:

register =>  73432260005:pass@10001
register =>  73432260050:pass@10002

[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap


Can you provide a sip debug of calls to both of these? I'm confused 
how that... works...





--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)





--
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Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Rainer Piper]

I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600

*extensions.conf* ; i'm cutting the dialed number out of the invite 
Header and goto/jump to the extensions

; incoming VOIP 9716716x SIPGATE
exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} *** 
${CALLERID(num)} ***)

same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same => n,NoOp( 49${gotoadr:-11} )
same => n,*Goto(49${gotoadr:-11},1)*

; the filter is jumping to the extensions ...

; incoming VOIP 97167160 SIPGATE -> MENU
exten => 
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r)

; incoming VOIP 97167161 SIPGATE
exten => 
4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r)

; incoming VOIP 97167162 SIPGATE ECHO TEST
exten => 
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167163 SIPGATE
exten => 
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167164 SIPGATE
exten => 
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167165 SIPGATE
exten => 
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incncoming VOIP 97167166 Mailbox
exten => 
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167167 CONF. 1
exten => 
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

; incoming VOIP 97167168 CONF. 2
;exten => 
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)

exten => 4922897167168,1,Answer
same => n,echo()
same => n,Hangup()
; incoming VOIP 97167169 FAX
;exten => 
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)



Regards
Rainer

Am 02.09.2014 um 15:08 schrieb Joshua Colp:

Nick Awesome wrote:

register =>  73432260005:pass@10001
register =>  73432260050:pass@10002

[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap


Can you provide a sip debug of calls to both of these? I'm confused 
how that... works...





--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
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Re: [asterisk-users] POSSA CID Superfecta to query a CardDAV server ?

[Lukasz Sokol]

On 02/09/14 09:12, Lukasz Sokol wrote:
> On 01/09/14 12:05, Marie Fischer wrote:
> 
>> Well, you made me curious - wrote up a little perl script to do a
>> filtered report by phone number. It takes 2-3 seconds to get a
>> response from OSX server (Mavericks). Which sure is shorter then
>> doing a full sync, but still longish. Would be interesting to know
>> how long other servers take.
> 
> I can try to time my CardDAV query program (keeping in mind that it
> runs on the same physical segment and subnet as the server) 
> (sorry I don't know Perl ;) I'm a Free Pascal user )
> Will come back to that.
> 

OK, alpha results of timing, show actually sending a query and receiving
a response takes ~500 - 800 ms. Total response time for a query 1.3~2.2s max,
if authentication required before query (but authentication request
can be sent by means of some cron sending OPTIONS & cookies every so often, 
to avoid session timeout; so counting that out, 500-800ms is what it takes 
for the server to answer the query).

>>
>> Now, for CID you would want this to run in your dial plan after the
>> call comes in and before you Dial() your local extension. One
>> ringtone is 5 seconds (1 sec tone, 4 sec silence), so it's actually
>> not too bad (remember those analog caller ID boxes which got the
>> caller ID between first and second ringtone?). Maybe you'd need to
>> send Progress() or Ringing() back to the calling party.
>>
> 
> Hmm ok.
> Let me time it and see the timings.
> 

Keeping in mind this is going to be running on FreePBX 2.11.0.38,
where do I start ;) if I wanted to marry this to CID Superfecta ?
Or should I rather look for how to write my own FreePBX module ?

You've mentioned 
'LWP::UserAgent and HTTP::Request and Text::vCard::Addressbook'
which I am going to research a bit.

The fun part would be to make the response from CardDAV into
an URL on which the operator shall click to bring the addressbook
details on screen 
(or just call firefox with url given as param that would do already ;) )

> Lukasz
> 
> 



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Re: [asterisk-users] Setup Own IP PBX Server

[Patrick Laimbock]

On 01-09-14 12:31, Chandran Manikandan wrote:
[snip]

I have installed Freepbx server and tried to configure sip extension.
It's working fine.


A better place for FreePBX related questions and to get help is:
http://community.freepbx.org/
Or hire their professional FreePBX support:
http://www.freepbx.org/support-and-professional-services

If you want to learn more about Asterisk in general then a good start is 
to first read "Asterisk: The Definitive Guide, 4th Edition" and go 
through the wiki at http://wiki.asterisk.org.


HTH,
Patrick

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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Joshua Colp]

Nick Awesome wrote:

register =>  73432260005:pass@10001
register =>  73432260050:pass@10002

[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap


Can you provide a sip debug of calls to both of these? I'm confused how 
that... works...


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

[Eric Wieling]

321 is not a valid Asterisk hangup cause.  Valid hangupcauses are 1-127 (Q.831 
cause codes)  See 
https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, September 02, 2014 12:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Custom SIP-header not present in call Asterisk to 
Asterisk


On 02-09-14 14:22, Eric Wieling wrote:
Try Hangup(123) where 123 is whatever "hangup cause" you want to send back to 
the caller.   The calliing Asterisk server will get the valuse back in 
HANGUPCAUSE variable.


Hello,

I have tried sending Hangup(321) on Asterisk server B to Asterisk A but when I 
read HangupCause on Asterisk A it always is '21'.

Good idea, but it does not seem to work.



Kind regards,

Jonas.
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Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

[Jonas Kellens]

On 02-09-14 14:22, Eric Wieling wrote:


Try Hangup(123) where 123 is whatever "hangup cause" you want to send 
back to the caller.   The calliing Asterisk server will get the valuse 
back in HANGUPCAUSE variable.





Hello,

I have tried sending Hangup(321) on Asterisk server B to Asterisk A but 
when I read HangupCause on Asterisk A it always is '21'.


Good idea, but it does not seem to work.



Kind regards,

Jonas.
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Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

[Eric Wieling]

Try Hangup(123) where 123 is whatever "hangup cause" you want to send back to 
the caller.   The calliing Asterisk server will get the valuse back in 
HANGUPCAUSE variable.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, September 02, 2014 8:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Custom SIP-header not present in call Asterisk to 
Asterisk

Hello,

I have a situation where a call comes in to my Asterisk server B. This call 
comes from another Asterisk server A. I want to tell to this server A why my 
server B hangs up.

So just before hanging up, I add a custom SIP-header :

exten => s,n,SIPAddHeader(X-My-Hangup: MaxChan)
exten => s,n,Hangup()


But I notice that this extra SIP-header is not send within the SIP-reponse :

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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Nick Awesome]

register => 73432260005:pass@10001
register => 73432260050:pass@10002

[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap


so now in context dialmap (agi application) AGI->agi_channel is 
'SIP/10001-0005’
parsing 10001 and checking db for matches, in db I have table with all my 
trunks information

On 02 Sep 2014, at 15:49, Joshua Colp  wrote:

> Nick Awesome wrote:
>> Tried doing that, but
>> 
>> first: AGI->exten is ’s’ for some reason. and second its not
>> practical, for example if 80.75.132.66 wound like to register on my *
>> server - it will not work because I already using that IP with
>> different endpoint
>> 
>> well, its critical trouble for me, coming back to chat_sip :|
> 
> How will you do this in chan_sip? The behavior between the two is the same, 
> despite the configuration being different.
> 
> -- 
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
> 
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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Joshua Colp]

Nick Awesome wrote:

Tried doing that, but

first: AGI->exten is ’s’ for some reason. and second its not
practical, for example if 80.75.132.66 wound like to register on my *
server - it will not work because I already using that IP with
different endpoint

well, its critical trouble for me, coming back to chat_sip :|


How will you do this in chan_sip? The behavior between the two is the 
same, despite the configuration being different.


--
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Nick Awesome]

Tried doing that, but

first: AGI->exten is ’s’ for some reason.
and second its not practical, for example if 80.75.132.66 wound like to 
register on my * server - it will not work because I already using that IP with 
different endpoint

well, its critical trouble for me, coming back to chat_sip :|

On 02 Sep 2014, at 15:32, A J Stiles  wrote:

> On Tuesday 02 Sep 2014, Nick Awesome wrote:
>> Hello guys.
>> 
>> Have 2 external numbers that required registration on provider server,
>> 
>> trunk1: 73432260005@80.75.132.66
>> trunk2: 73432260050@80.75.132.66
>> 
>> Thing is I can’t figure out how to route them to different IVRs
>> 
>> by default Asterisk can’t match endpoint
>> 
>> Request from '' failed for
>> '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
>> matching endpoint found
>> 
>> Can’t set identify by IP because they got the same ip.
>> 
>> Is there way to configure asterisk so incoming calls from same IP but
>> different ID will use different contexts?
> 
> Can't you send them both to the same context initially; but once you are 
> there, match the outside number  (which can be found in ${EXTEN} if it is the 
> number that was dialled from their end, or ${CALLERID(num)} if it is the 
> number they are calling from)  within that context and use a GoToIf() to send 
> calls from trunk 2 to the correct context?
> 
> -- 
> AJS
> 
> Note:  Originating address only accepts e-mail from list!  If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
> 
> -- 
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Re: [asterisk-users] PJSIP issues with handling incoming calls

[A J Stiles]

On Tuesday 02 Sep 2014, Nick Awesome wrote:
> Hello guys.
> 
> Have 2 external numbers that required registration on provider server,
> 
> trunk1: 73432260005@80.75.132.66
> trunk2: 73432260050@80.75.132.66
> 
> Thing is I can’t figure out how to route them to different IVRs
> 
> by default Asterisk can’t match endpoint
> 
> Request from '' failed for
> '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
> matching endpoint found
> 
> Can’t set identify by IP because they got the same ip.
> 
> Is there way to configure asterisk so incoming calls from same IP but
> different ID will use different contexts?

Can't you send them both to the same context initially; but once you are 
there, match the outside number  (which can be found in ${EXTEN} if it is the 
number that was dialled from their end, or ${CALLERID(num)} if it is the 
number they are calling from)  within that context and use a GoToIf() to send 
calls from trunk 2 to the correct context?

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Joshua Colp]

Nick Awesome wrote:

Thats because I call from one to other


Then no, you can only match based on IP address. This also applies to 
chan_sip. You have to send both to the same context and then within 
there you can differentiate them based on the dialed number.


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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Nick Awesome]

Thats because I call from one to other

here’s logs where I call from mobile

<--- Received SIP request (469 bytes) from UDP:80.75.132.66:5060 --->
ACK sip:s@pbx_ip_address:57408;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 
80.75.132.66:5060;branch=z9hG4bK-524287-1-NGJlZjMxNThhYjI2YjI3Y2EyODE0MThhMTVkNjY0ZTA.--5c6da819e6300b26;rport
Max-Forwards: 70
To: 
;tag=z9hG4bK-524287-1-NGJlZjMxNThhYjI2YjI3Y2EyODE0MThhMTVkNjY0ZTA.--5c6da819e6300b26
From: ;tag=7ozmpvsvqs26kcor.o
Call-ID: 18e2786560719216837824k41099rmwp
CSeq: 586 ACK
Content-Length: 0


<--- Received SIP request (469 bytes) from UDP:80.75.132.66:5060 --->
ACK sip:s@pbx_ip_address:57408;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 
80.75.132.66:5060;branch=z9hG4bK-524287-1-YzhmMDE2NzE1YjRhOTM4NjQ1MjMxMmMyMmM0MWFiZTE.--be7c48325cdef400;rport
Max-Forwards: 70
To: 
;tag=z9hG4bK-524287-1-YzhmMDE2NzE1YjRhOTM4NjQ1MjMxMmMyMmM0MWFiZTE.--be7c48325cdef400
From: ;tag=yddmzvcoi3waw24e.o
Call-ID: 22e7064301970213226722k41100rmwp
CSeq: 588 ACK
Content-Length: 0

On 02 Sep 2014, at 15:01, Joshua Colp  wrote:

> Nick Awesome wrote:
>> Hello guys.
> 
> Kia ora,
> 
>> Have 2 external numbers that required registration on provider server,
>> 
>> trunk1: 734322600*05*@80.75.132.66
>> trunk2: 734322600*50*@80.75.132.66
>> 
>> Thing is I can’t figure out how to route them to different IVRs
>> 
>> by default Asterisk can’t match endpoint
>> 
>> Request from '' failed for
>> '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
>> matching endpoint found
>> 
>> Can’t set /identify /by IP because they got the same ip.
>> 
>> Is there way to configure asterisk so incoming calls from same IP but
>> different ID will use different contexts?
> 
> If the From header contains the destination number (as it seems to based on 
> your above log message and config) you can create two different endpoints and 
> match based on the user portion of the From header.
> 
> [734322600*05*]
> type=endpoint
> context=did-1
> disallow=all
> allow=ulaw
> 
> [734322600*50*]
> type=endpoint
> context=did-2
> disallow=all
> allow=ulaw
> 
> If this is not correct then you can only match once based on the source IP 
> address currently.
> 
> Cheers,
> 
> -- 
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
> 
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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Joshua Colp]

Nick Awesome wrote:

Hello guys.


Kia ora,


Have 2 external numbers that required registration on provider server,

trunk1: 734322600*05*@80.75.132.66
trunk2: 734322600*50*@80.75.132.66

Thing is I can’t figure out how to route them to different IVRs

by default Asterisk can’t match endpoint

Request from '' failed for
'80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
matching endpoint found

Can’t set /identify /by IP because they got the same ip.

Is there way to configure asterisk so incoming calls from same IP but
different ID will use different contexts?


If the From header contains the destination number (as it seems to based 
on your above log message and config) you can create two different 
endpoints and match based on the user portion of the From header.


[734322600*05*]
type=endpoint
context=did-1
disallow=all
allow=ulaw

[734322600*50*]
type=endpoint
context=did-2
disallow=all
allow=ulaw

If this is not correct then you can only match once based on the source 
IP address currently.


Cheers,

--
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

[A J Stiles]

On Tuesday 02 Sep 2014, Jonas Kellens wrote:
> On 02-09-14 11:34, Steven Howes wrote:
> > On 2 Sep 2014, at 09:03, Jonas Kellens  > 
> > > wrote:
> >> So just before hanging up, I add a custom SIP-header :
> >> 
> >> exten => s,n,SIPAddHeader(X-My-Hangup: MaxChan)
> >> exten => s,n,Hangup()
> > 
> > SIPAddHeader only works for INVITE as far as I know.
> > 
> > Steve
> 
> OK.
> 
> Then how can I let another Asterisk server know the custom reason of
> hangup ? If it is not possible with custom SIP-header, then how ?

Fire off an AGI script which will  (somehow)  send the necessary message to the 
other Asterisk server.

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Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

[Steven Howes]

On 2 Sep 2014, at 10:38, Jonas Kellens  wrote:
> Then how can I let another Asterisk server know the custom reason of hangup ? 
> If it is not possible with custom SIP-header, then how ?

As far as I know that’s going to require a source change. May not be the case 
with PJSIP though - not used that yet.

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Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

[Jonas Kellens]

On 02-09-14 11:34, Steven Howes wrote:
On 2 Sep 2014, at 09:03, Jonas Kellens > wrote:

So just before hanging up, I add a custom SIP-header :

exten => s,n,SIPAddHeader(X-My-Hangup: MaxChan)
exten => s,n,Hangup()


SIPAddHeader only works for INVITE as far as I know.

Steve


OK.

Then how can I let another Asterisk server know the custom reason of 
hangup ? If it is not possible with custom SIP-header, then how ?




Regards,

Jonas.
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Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

[Steven Howes]

On 2 Sep 2014, at 09:03, Jonas Kellens  wrote:
> So just before hanging up, I add a custom SIP-header :
> 
> exten => s,n,SIPAddHeader(X-My-Hangup: MaxChan)
> exten => s,n,Hangup()

SIPAddHeader only works for INVITE as far as I know.

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Re: [asterisk-users] POSSA CID Superfecta to query a CardDAV server ?

[Lukasz Sokol]

On 01/09/14 12:05, Marie Fischer wrote:

> Well, you made me curious - wrote up a little perl script to do a
> filtered report by phone number. It takes 2-3 seconds to get a
> response from OSX server (Mavericks). Which sure is shorter then
> doing a full sync, but still longish. Would be interesting to know
> how long other servers take.

I can try to time my CardDAV query program (keeping in mind that it
runs on the same physical segment and subnet as the server) 
(sorry I don't know Perl ;) I'm a Free Pascal user )
Will come back to that.

> 
> Now, for CID you would want this to run in your dial plan after the
> call comes in and before you Dial() your local extension. One
> ringtone is 5 seconds (1 sec tone, 4 sec silence), so it's actually
> not too bad (remember those analog caller ID boxes which got the
> caller ID between first and second ringtone?). Maybe you'd need to
> send Progress() or Ringing() back to the calling party.
> 

Hmm ok.
Let me time it and see the timings.

Lukasz


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[asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

[Jonas Kellens]

Hello,

I have a situation where a call comes in to my Asterisk server B. This 
call comes from another Asterisk server A. I want to tell to this server 
A why my server B hangs up.


So just before hanging up, I add a custom SIP-header :

exten => s,n,SIPAddHeader(X-My-Hangup: MaxChan)
exten => s,n,Hangup()


But I notice that this extra SIP-header is not send within the SIP-reponse :

SIP/2.0 603 Declined
Via: SIP/2.0/UDP 
xx.xx.xx.98:5060;branch=z9hG4bK168884d7;received=xx.xx.xx.98;rport=5060

From: "5006" ;tag=as50c98b4c
To: ;tag=as3c6e57b0
Call-ID: 6d1039bb22716c6e6dec69fb3e78a...@xx.xx.xx.98:5060
CSeq: 102 INVITE
Server: myasterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH

Supported: replaces, timer
Content-Length: 0


How can I make this work ?


Thanks.

Jonas.
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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Administrator TOOTAI]

Le 02/09/2014 09:38, Nick Awesome a écrit :

So there is no way to do that with pjsip?


Sorry, I didn't read carefully the subject. I can't answer for pjsip. My 
bad :-(




On 02 Sep 2014, at 11:35, Administrator TOOTAI  wrote:


Le 02/09/2014 08:47, Nick Awesome a écrit :

Hello guys.


Hi



Have 2 external numbers that required registration on provider server,

trunk1: 734322600*05*@80.75.132.66
trunk2: 734322600*50*@80.75.132.66

Thing is I can’t figure out how to route them to different IVRs

by default Asterisk can’t match endpoint

Request from '' failed for
'80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
matching endpoint found

Can’t set /identify /by IP because they got the same ip.

Is there way to configure asterisk so incoming calls from same IP but
different ID will use different contexts?


You have to register to the gateway with each account user and password like

sip.conf

register = 734322600*05*:password1@myProvider/734322600*05*
register = 734322600*50*:password2@myProvider/734322600*50*

[myProvider]
type=peer
host=80.75.132.66
context=from-myProvider
...

extensions.conf

[from-myProvider]
exten = 734322600*05*,1,NoOp(Incoming call to 734322600*05*)
...

exten = 734322600*50*,1,NoOp(Incoming call to 734322600*50*)
...

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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Nick Awesome]

So there is no way to do that with pjsip?

On 02 Sep 2014, at 11:35, Administrator TOOTAI  wrote:

> Le 02/09/2014 08:47, Nick Awesome a écrit :
>> Hello guys.
> 
> Hi
> 
>> 
>> Have 2 external numbers that required registration on provider server,
>> 
>> trunk1: 734322600*05*@80.75.132.66
>> trunk2: 734322600*50*@80.75.132.66
>> 
>> Thing is I can’t figure out how to route them to different IVRs
>> 
>> by default Asterisk can’t match endpoint
>> 
>> Request from '' failed for
>> '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
>> matching endpoint found
>> 
>> Can’t set /identify /by IP because they got the same ip.
>> 
>> Is there way to configure asterisk so incoming calls from same IP but
>> different ID will use different contexts?
> 
> You have to register to the gateway with each account user and password like
> 
> sip.conf
> 
> register = 734322600*05*:password1@myProvider/734322600*05*
> register = 734322600*50*:password2@myProvider/734322600*50*
> 
> [myProvider]
> type=peer
> host=80.75.132.66
> context=from-myProvider
> ...
> 
> extensions.conf
> 
> [from-myProvider]
> exten = 734322600*05*,1,NoOp(Incoming call to 734322600*05*)
> ...
> 
> exten = 734322600*50*,1,NoOp(Incoming call to 734322600*50*)
> ...
> 
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Re: [asterisk-users] AGI scripts - delay issue.

[Thorsten Göllner]

Am 02.09.2014 07:09, schrieb Bryant Zimmerman:

Hey All
We have several AGI scripts that access databases. These work well 
most of the time.
The issue we are having is that on rare occasion our script must fail 
to a backup database server.
When this occurs it may take up to two seconds to do so. The issue is 
when there is this delay the script loses access to read global 
channel  variable values only after the delay. This is driving me 
crazy is there some kind of  AGI timeout issue or bug that could be 
causing this.


What do you mean with "the script loses access to read global channel  
variable values"? What is the asterisk version? What channel tech is 
used? What type of AGI-Scripts do you use?
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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Administrator TOOTAI]

Le 02/09/2014 08:47, Nick Awesome a écrit :

Hello guys.


Hi



Have 2 external numbers that required registration on provider server,

trunk1: 734322600*05*@80.75.132.66
trunk2: 734322600*50*@80.75.132.66

Thing is I can’t figure out how to route them to different IVRs

by default Asterisk can’t match endpoint

Request from '' failed for
'80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
matching endpoint found

Can’t set /identify /by IP because they got the same ip.

Is there way to configure asterisk so incoming calls from same IP but
different ID will use different contexts?


You have to register to the gateway with each account user and password like

sip.conf

register = 734322600*05*:password1@myProvider/734322600*05*
register = 734322600*50*:password2@myProvider/734322600*50*

[myProvider]
type=peer
host=80.75.132.66
context=from-myProvider
...

extensions.conf

[from-myProvider]
exten = 734322600*05*,1,NoOp(Incoming call to 734322600*05*)
...

exten = 734322600*50*,1,NoOp(Incoming call to 734322600*50*)
...

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