Re: [asterisk-users] cli command show codecs

[Bruce B]

if you see leg-A as ulaw and leg-B as g729 then it's trans-coding. If your
IVR announcement is not recorded in g729 and you see g729 on the channel
when you call into IVR then it's transcoding as well.

On Wed, Aug 31, 2011 at 5:50 PM, Eric Wieling  wrote:

> Assuming SIP "sip show channels" will show you which codec is used for each
> call leg.  However it does not track transcoding.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
> Sent: Wednesday, August 31, 2011 2:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] cli command show codecs
>
> asterisk -rx "core show channels verbose" does not provide transcoding
> details.
>
> Unless I have missed something.
>
> Sans
>
>
>
> On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas 
> wrote:
>
>
>Core show channels verbose is probably your best bet.  I think the
> answer also depends on your * version.
>
>
>
>From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
>Sent: Wednesday, August 31, 2011 10:44 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [asterisk-users] cli command show codecs
>
>
>
>Hi,
>
>Is there a CLI command which will tell me the codec used for active
> calls and if transcoding is happening ?
>
>Thx
>Sans
>
>
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Re: [asterisk-users] Linksys/Cisco 504G randomly restarts

[C F]

On Tue, Aug 16, 2011 at 10:27 AM, Jose Flores Galicia  wrote:
> I have seen this with last firmware.
>
> You need to change those 2 parameters to get a non-autoreboot scenario:
>
>   0001
>   

Thanks, looks like this did the trick.


>
> Whenever it resync profile, it reboots, so setting resync to an hour nobody
> uses it and unseting resync periodicaly solve it for me.
>
> Jose Flores Galicia
> BriefCode && Code Based Training
>
>
> 2011/8/15 C F 
>>
>> On Mon, Aug 15, 2011 at 2:54 PM, Vahan Yerkanian 
>> wrote:
>> > On 8/15/11 10:46 PM, C F wrote:
>> >>
>> >> I have 3 Linksys/Cisco 504G phones they keep restarting at what seems
>> >> to be random. Sometimes as short as 6 minutes.
>> >> FW version is 7.4.3a
>> >>
>> >> I have searched and tried disabling FW check and all related settings.
>> >> I also extended all the default 3600 resync checks to a lot longer.
>> >>
>> >> TIA
>> >> CF
>> >
>> > Hi,
>> >
>> > Try upgrading to the latest version. I have tens of 504G operating
>> > without
>> > any problems.
>>
>> That is going to be my next step. Thank you
>>
>> >
>> > How are you powering these phones? I had a case when a PoE switch was
>> > experiencing short-circuit problems on a badly wired cable, and was
>> > unable
>> > to provide enough current to power the phones on the other ports.
>> > Replacing
>> > the faulty cable fixed the problem. You can always try to power the
>> > phones
>> > using 5volts DC, 2A center pin positive power source and see if the
>> > problem
>> > persists.
>>
>> They are all locally powered. I can see in asterisk CLI that its not a
>> power issue since they are unregistering before shutdown.
>>
>> Thanks again.
>>
>> >
>> > Also I have a Linksys SPA-941 that has a public IP and reboots itself
>> > whenever someone tries to bruteforce into it by sending tons of sip
>> > registers :)
>> >
>> > HTH,
>> > Vahan
>> >
>> >
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Re: [asterisk-users] Wanted a modified SIP message body

[Larry Moore]

On 31/08/2011 11:23 PM, Olle E. Johansson wrote:

31 aug 2011 kl. 14:42 skrev Kevin P. Fleming:


On 08/31/2011 02:46 AM, Jaime Lozano wrote:

Hello,
I agree with you, I'm not explaining the problem in a proper manner,
because of my lack of Asterisk knowings. I send the Wireshark captures.

3com telephones take the timezone TZ:7200 from the 3Com PBX to show the
time right. But what if I want a 3Com telephone to work with Asterisk
PBX? Then the telephone time is wrong, 2 hours lower. It seems 3Com
telephones need the TZ:7200. 3Com telephones work with Asterisk and it
is great, but we would like to log the calls.

OK, so the first clarification is that you are talking about responses to 
REGISTER requests specifically, not all responses to all requests. That's good 
:-)

On to the meat of the issue... indeed, the '200 OK' response to a REGISTER 
request does not normally have a message body; nothing in the SIP RFCs even 
suggests that there would be one (although it's certainly allowed should the 
registrar want to include it) or what would be present in it.

As has been previously replied here, there is no facility in Asterisk to 
include a message body in a REGISTER request response, so providing one will 
definitely require source code modifications. They wouldn't be terribly 
difficult, but they would only be applicable to these particular phones, which 
reduces the benefit of making the changes to the community at large.

With that said... it's certainly possible to do this, but it's going to take 
some non-trivial code changes. The REGISTER handling code does not use any of 
the methods that exist in chan_sip to add message body content to its 
responses, it uses simpler methods that assume there won't be a message body.

In addition, this mechanism is really pretty broken anyway; the server would 
have to know where each phone is physically located in order to be able to 
provide the correct TZ value to it, and would have to be updated if the phone 
is moved. Not an ideal situation.

The RFC states that a phone could use the Date: header in the response to set 
the local time in the device. It's always in GMT which makes it stupid to add a 
time zone any where.

-1 for this implementation.


Perhaps researching the specs./capabilities of the phone for other 
capabilities setting its time zone. A DHCP server can offer a 
time-offset value, whether the phone can be provisioned with a defined  
time zone offset or accept the offset in DHCP is a matter of further 
research.


Larry.

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Re: [asterisk-users] Avaya to Asterisk Voice mail

[C F]

What exactly is your setup?

On Tue, Aug 30, 2011 at 10:44 AM, Dustin fails  wrote:
> Has anyone have Avaya setup to ring to Asterisk voice mail over an analogue
> line. The issue I am having is Avaya is sending the originating caller id
> not the station id so Asterisk see that originating id so I can't route the
> call correctly in Asterisk.
>
> Thanks!
>
> Dustin
>
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Re: [asterisk-users] Dragging the dialup customers along, possible?

[C F]

On Wed, Aug 31, 2011 at 9:25 AM, Kevin P. Fleming  wrote:
> On 08/29/2011 10:32 PM, C F wrote:
>>
>> On Mon, Aug 29, 2011 at 4:32 PM, Cary Fitch  wrote:
>>>
>>>
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com
>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
>>> Sent: Monday, August 29, 2011 3:18 PM
>>> To: t...@ewebforce.net; Asterisk Users Mailing List - Non-Commercial
>>> Discussion
>>> Subject: Re: [asterisk-users] Dragging the dialup customers along,
>>> possible?
>>>
>>>  From what you are asking it appears that you are trying to run similar
>>> to a fax (modulation and demodulation) over VoIP.
>>> Try again, the fact that you succeeded twice was pure luck, and as far
>>> as I understand that didn't even work out.
>>> Switch back to TDM. Your dial up modems want that magic thing called
>>> timing and no jitter that only TDM will give you.
>>>
>>> ===
>>>

>> So you want to develop the equivalent of T.38 for dial up?
>
> It already exists; it's called V.150.

Wow, thanks for the info.

>
> --
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Re: [asterisk-users] Faxes suddenly failing

[C F]

I think you should change the subject line to:
Faxes suddenly worked for 2 weeks.

On Wed, Aug 31, 2011 at 3:49 PM, Tim King  wrote:
> I realize that faxing is not great with voip but here is my confusion. I
> have been working on a web based fax system for 2 weeks. During this time I
> have sent over 100 2 page faxes without any errors. Now today as things are
> finally completed I can not seem to get any fax to go through unless it is a
> 1 page cover only. Anyone able to tell the issue from this debug output?
>
>    -- Channel 'SIP/MyVoipProvider-0046' FAX session '12' started
>     -- FAX handle 0: [ 000.38 ], STAT_EVT_STRT_RX   st: IDLE
> rt: IDLENSRX
>     -- FAX handle 0: [ 000.000184 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY
> rt: RRDYNHRY
>     -- FAX handle 0: [ 000.000504 ], P30EVN_RECEIVE_STARTED
>     -- FAX handle 0: [ 000.000538 ], STAT_INFO_CSI
>     -- FAX handle 0: [ 000.000568 ], STAT_INFO_DIS
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 000.091837 ], stack sent 5 frames (100 ms) of energy.
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 000.160248 ], stack sent 3 frames (60 ms) of silence.
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 000.960201 ], channel sent 48 frames (960 ms) of silence.
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 000.979464 ], channel sent 1 frames (20 ms) of energy.
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 003.157848 ], stack sent 150 frames (3000 ms) of energy.
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 003.219814 ], stack sent 3 frames (60 ms) of silence.
>     -- FAX handle 0: [ 005.240927 ], STAT_EVT_TX_V21_DONE   st: WT_DIS_RSP
> rt: WDSRNT21
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 005.579811 ], stack sent 118 frames (2360 ms) of energy.
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 006.481179 ], channel sent 275 frames (5500 ms) of silence.
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 007.801045 ], channel sent 66 frames (1320 ms) of energy.
>     -- FAX handle 0: [ 007.800554 ], STAT_FRM_CRP
>     -- FAX handle 0: [ 007.800586 ], STAT_EVT_CRP   st: WT_DIS_RSP
> rt: NT4X
>     -- FAX handle 0: [ 007.800602 ], STAT_EVT_FSC_ERR   st: WT_DIS_RSP
> rt: UNEXPECT
>     -- FAX handle 0: [ 011.012832 ], STAT_EVT_RX_TRN_END    st: WT_DIS_RSP
> rt: RXXXNFRX
>     -- FAX handle 0: [ 011.012878 ], STAT_INFO_CSI
>     -- FAX handle 0: [ 011.012905 ], STAT_INFO_DIS
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 011.152812 ], stack sent 279 frames (5580 ms) of silence.
>     -- FAX handle 0: [ 013.179561 ], STAT_EVT_TX_V21_DONE   st: WT_DIS_RSP
> rt: WDSRNT21
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 013.471827 ], stack sent 116 frames (2320 ms) of energy.
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 014.260642 ], channel sent 323 frames (6460 ms) of silence.
>     -- FAX handle 0: [ 016.119786 ], STAT_INFO_TSI
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 016.460661 ], channel sent 110 frames (2200 ms) of energy.
>     -- FAX handle 0: [ 016.460394 ], STAT_INFO_DCS
>     -- FAX handle 0: [ 016.460431 ], STAT_EVT_DCS   st: WT_DIS_RSP
> rt: WDSRNDCS
>     -- FAX handle 0: [ 016.460449 ], STAT_NEG_V17_14400
>     -- FAX handle 0: [ 016.460464 ], STAT_NEG_MH
>     -- FAX handle 0: [ 016.460476 ], STAT_NEG_A4
>     -- FAX handle 0: [ 016.460488 ], STAT_NEG_RES_204x196
>     -- FAX handle 0: [ 016.460500 ], STAT_NEG_ECM
>     -- FAX handle 0: [ 016.460514 ], STAT_EVT_SW_ECM    st: WT_DIS_RSP
> rt: WDSRNSWE
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 016.540315 ], channel sent 4 frames (80 ms) of silence.
>     -- FAX handle 0: [ 016.800906 ], STAT_EVT_RX_IMG_STRT   st: RCV_ECM_TRN
> rt: UNEXPECT
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 019.700543 ], channel sent 158 frames (3160 ms) of energy.
>     -- FAX handle 0: [ 019.759984 ], STAT_EVT_RX_TRN_END    st: RCV_ECM_TRN
> rt: RTCFNERT
>     -- FAX handle 0: [ 019.760071 ], STAT_FRM_CFR
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 019.912812 ], stack sent 322 frames (6440 ms) of silence.
>     -- FAX handle 0: [ 020.957834 ], STAT_EVT_TX_V21_DONE   st: RCV_ECM_STRT
> rt: RECMNT21
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 021.278809 ], stack sent 68 frames (1360 ms) of energy.
>    > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
> 022.261160 ], channel sent 128 frames (2560 ms) of silence.
>     -- FAX handle 0: [ 022.517880 ], STAT_EVT_RX_IMG_STRT   st: RCV_ECM_STRT
> rt: RECMNSRI
>     -- FAX handle 0: [ 022.517982 ], P30EVN_PHASE_C
>     -- FAX handle 0: [ 022.517998 ], P30EVN_DOC_START
>     -- FAX handle 0: [ 022.518429 

Re: [asterisk-users] cli command show codecs

[Eric Wieling]

Assuming SIP "sip show channels" will show you which codec is used for each 
call leg.  However it does not track transcoding.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Wednesday, August 31, 2011 2:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] cli command show codecs

asterisk -rx "core show channels verbose" does not provide transcoding details.

Unless I have missed something.

Sans



On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas  wrote:


Core show channels verbose is probably your best bet.  I think the 
answer also depends on your * version.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Wednesday, August 31, 2011 10:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] cli command show codecs

 

Hi,

Is there a CLI command which will tell me the codec used for active 
calls and if transcoding is happening ?

Thx
Sans


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[asterisk-users] phone + video

[Hans Witvliet]

Hi all,

I know that a lot of people have negative experiences with
grandstream-2000, but personally. i'd only the repace one poweradapter
after three years...

So, can anybody give some comment on one of their recent models,
the GXV-3175 (the one with the 7" display)
I'm looking for a phone with video capabilities, as i don;t want to
limit my self to testing with softphone..

HtH, Hans

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Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system

[Hans Witvliet]

On Fri, 2011-08-26 at 19:03 -0400, Eric Wieling wrote:
> >-Original Message-
> >From: asterisk-users-boun...@lists.digium.com 
> >[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
> >Sent: Friday, August 26, 2011 6:09 PM
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
> >Subject: Re: [asterisk-users] Looking for ideas for nice **Asterisk** home 
> >phone system
> >
> >On Fri, 2011-08-26 at 07:41 -0700, Steve Edwards wrote:
> >> On Fri, 26 Aug 2011, linux guy wrote:
> >> 
> > > How much power does the home asterisk box need ? 
> >
> >I use a small box (like those hp thin clients) But these are a bit stronger 
> >aluminium housing, instead of plastic, and better foor cooling.
> >
> >Power consumption: 8 Watt under full load
> >CPU:  Model: 6.28.2 "Intel(R) Atom(TM) CPU Z530   @ 1.60GHz"
> >Memory Size: 1 GB
> >Disk /dev/sda: 64.0 GB, 64023257088 bytes This model has just one ethernet 
> >port, others have two
> >Size: 10x10 cm
> 
> Is this a custom build box or does a company sell them preassembled?We 
> are always on the lookout for potential boxes we can use for small 
> installations.
> 
It is pre-assembled,
You can opt for either no internal disk small (8GB) sdd, larger (64GB)
sdd or ordinary disk, And either no, MS, or ubuntu pre-installed.
Also with/without wifi antenna.

hw

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[asterisk-users] Faxes suddenly failing

[Tim King]

I realize that faxing is not great with voip but here is my confusion. I
have been working on a web based fax system for 2 weeks. During this time I
have sent over 100 2 page faxes without any errors. Now today as things are
finally completed I can not seem to get any fax to go through unless it is a
1 page cover only. Anyone able to tell the issue from this debug output?

   -- Channel 'SIP/MyVoipProvider-0046' FAX session '12' started
-- FAX handle 0: [ 000.38 ], STAT_EVT_STRT_RX   st: IDLE
rt: IDLENSRX
-- FAX handle 0: [ 000.000184 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY
rt: RRDYNHRY
-- FAX handle 0: [ 000.000504 ], P30EVN_RECEIVE_STARTED
-- FAX handle 0: [ 000.000538 ], STAT_INFO_CSI
-- FAX handle 0: [ 000.000568 ], STAT_INFO_DIS
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
000.091837 ], stack sent 5 frames (100 ms) of energy.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
000.160248 ], stack sent 3 frames (60 ms) of silence.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
000.960201 ], channel sent 48 frames (960 ms) of silence.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
000.979464 ], channel sent 1 frames (20 ms) of energy.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
003.157848 ], stack sent 150 frames (3000 ms) of energy.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
003.219814 ], stack sent 3 frames (60 ms) of silence.
-- FAX handle 0: [ 005.240927 ], STAT_EVT_TX_V21_DONE   st: WT_DIS_RSP
rt: WDSRNT21
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
005.579811 ], stack sent 118 frames (2360 ms) of energy.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
006.481179 ], channel sent 275 frames (5500 ms) of silence.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
007.801045 ], channel sent 66 frames (1320 ms) of energy.
-- FAX handle 0: [ 007.800554 ], STAT_FRM_CRP
-- FAX handle 0: [ 007.800586 ], STAT_EVT_CRP   st: WT_DIS_RSP
rt: NT4X
-- FAX handle 0: [ 007.800602 ], STAT_EVT_FSC_ERR   st: WT_DIS_RSP
rt: UNEXPECT
-- FAX handle 0: [ 011.012832 ], STAT_EVT_RX_TRN_ENDst: WT_DIS_RSP
rt: RXXXNFRX
-- FAX handle 0: [ 011.012878 ], STAT_INFO_CSI
-- FAX handle 0: [ 011.012905 ], STAT_INFO_DIS
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
011.152812 ], stack sent 279 frames (5580 ms) of silence.
-- FAX handle 0: [ 013.179561 ], STAT_EVT_TX_V21_DONE   st: WT_DIS_RSP
rt: WDSRNT21
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
013.471827 ], stack sent 116 frames (2320 ms) of energy.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
014.260642 ], channel sent 323 frames (6460 ms) of silence.
-- FAX handle 0: [ 016.119786 ], STAT_INFO_TSI
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
016.460661 ], channel sent 110 frames (2200 ms) of energy.
-- FAX handle 0: [ 016.460394 ], STAT_INFO_DCS
-- FAX handle 0: [ 016.460431 ], STAT_EVT_DCS   st: WT_DIS_RSP
rt: WDSRNDCS
-- FAX handle 0: [ 016.460449 ], STAT_NEG_V17_14400
-- FAX handle 0: [ 016.460464 ], STAT_NEG_MH
-- FAX handle 0: [ 016.460476 ], STAT_NEG_A4
-- FAX handle 0: [ 016.460488 ], STAT_NEG_RES_204x196
-- FAX handle 0: [ 016.460500 ], STAT_NEG_ECM
-- FAX handle 0: [ 016.460514 ], STAT_EVT_SW_ECMst: WT_DIS_RSP
rt: WDSRNSWE
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
016.540315 ], channel sent 4 frames (80 ms) of silence.
-- FAX handle 0: [ 016.800906 ], STAT_EVT_RX_IMG_STRT   st: RCV_ECM_TRN
rt: UNEXPECT
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
019.700543 ], channel sent 158 frames (3160 ms) of energy.
-- FAX handle 0: [ 019.759984 ], STAT_EVT_RX_TRN_ENDst: RCV_ECM_TRN
rt: RTCFNERT
-- FAX handle 0: [ 019.760071 ], STAT_FRM_CFR
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
019.912812 ], stack sent 322 frames (6440 ms) of silence.
-- FAX handle 0: [ 020.957834 ], STAT_EVT_TX_V21_DONE   st: RCV_ECM_STRT
rt: RECMNT21
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
021.278809 ], stack sent 68 frames (1360 ms) of energy.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
022.261160 ], channel sent 128 frames (2560 ms) of silence.
-- FAX handle 0: [ 022.517880 ], STAT_EVT_RX_IMG_STRT   st: RCV_ECM_STRT
rt: RECMNSRI
-- FAX handle 0: [ 022.517982 ], P30EVN_PHASE_C
-- FAX handle 0: [ 022.517998 ], P30EVN_DOC_START
-- FAX handle 0: [ 022.518429 ], P30EVN_PAGE_START
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
031.102000 ], channel sent 442 frames (8840 ms) of energy.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
031.160415 ], channel sent 3 frames (60 ms) of silence.
-- FAX handle 0: [ 031.160196 

Re: [asterisk-users] T.38 passthru on 1.8.5

[Kevin P. Fleming]

On 08/31/2011 09:43 AM, Fabian Borot wrote:

Hi Kevin, I created the issue on the https://issues.asterisk.org/jira
web site, posted the description of the prob and submitted asterisk
console logs [sip and udptl debug on] and a wireshark capture taken on
the asterisk machine showing both legs with signaling and media.
PLease let me know what other thing you need you need me to provide.


I've already started looking at the packet capture; I'll follow up on 
the issue itself (ASTERISK-18394 for those following along at home).


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] cli command show codecs

[Bruce B]

"sip show channels" is the command you are looking for.

On Wed, Aug 31, 2011 at 2:45 PM, RSCL Mumbai  wrote:

> asterisk -rx "core show channels verbose" does not provide transcoding
> details.
>
> Unless I have missed something.
>
> Sans
>
>
> On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas wrote:
>
>> Core show channels verbose is probably your best bet.  I think the answer
>> also depends on your * version.
>>
>> ** **
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *RSCL Mumbai
>> *Sent:* Wednesday, August 31, 2011 10:44 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* [asterisk-users] cli command show codecs
>>
>> ** **
>>
>> Hi,
>>
>> Is there a CLI command which will tell me the codec used for active calls
>> and if transcoding is happening ?
>>
>> Thx
>> Sans
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>
>
>
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Re: [asterisk-users] cli command show codecs

[RSCL Mumbai]

asterisk -rx "core show channels verbose" does not provide transcoding
details.

Unless I have missed something.

Sans


On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas  wrote:

> Core show channels verbose is probably your best bet.  I think the answer
> also depends on your * version.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *RSCL Mumbai
> *Sent:* Wednesday, August 31, 2011 10:44 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] cli command show codecs
>
> ** **
>
> Hi,
>
> Is there a CLI command which will tell me the codec used for active calls
> and if transcoding is happening ?
>
> Thx
> Sans
>
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> _
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[asterisk-users] Asterisk 1.8.6.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team announces the release of Asterisk 1.8.6.0. 
This

release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.6.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* Fix an issue with Music on Hold classes losing files in playlist when 
realtime

  is used.
  (Closes issue ASTERISK-17875. Reported by David Cunningham. Patched 
by Igor

  Goncharovsky)

* Resolve a potential crash in chan_sip when utilizing auth= and 
performing a

  'sip reload' from the console.
  (Closes issue ASTERISK-17939. Reported by wdoekes. Patched by Richard 
Mudgett)


* Address some improper sql statements in res_odbc that would cause an 
update

  to fail on realtime peers due to trying to set as "(NULL)" rather than an
  actual NULL.
  (Closes issue ASTERISK-17791. Reported by marcelloceschia. Patched by 
Tilghman

  Lesher)

* Resolve issue where 403 Forbidden would always be sent maximum number 
of times

  regardless to receipt of ACK.
  (Patched by Richard Mudgett)

* Resolve issue where if a call to MeetMe includes both the dynamic(D) and
  always request PIN(P) options, MeetMe will ask for the PIN two times: 
once for

  creating the conference and once for entering the conference.
  (Patched by Kinsey Moore)

* Fix New Zealand indications profile based on
  http://www.telepermit.co.nz/TNA102.pdf
  (Closes issue ASTERISK-16263. Reported, Patched by richardf)

* Segfault in shell_helper in func_shell.c
  (Closes issue ASTERISK-18109. Reported by Michael Myles, patched by 
Richard

  Mudgett)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.6.0

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] cli command show codecs

[Danny Nicholas]

Core show channels verbose is probably your best bet.  I think the answer also 
depends on your * version.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Wednesday, August 31, 2011 10:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] cli command show codecs

 

Hi,

Is there a CLI command which will tell me the codec used for active calls and 
if transcoding is happening ?

Thx
Sans

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Re: [asterisk-users] USB or Ethernet based FXO device ?

[Andrew Latham]

On Wed, Aug 31, 2011 at 11:49 AM, Carlos Chavez  wrote:
> On Wed, 2011-08-31 at 17:03 +0200, Marco Signorini wrote:
>> Hi.
>>
>> I was following this thread. We normally use Patton SmartNode SN4112
>> series to interface to FXO ports. But I'm looking for something
>> different for a future setup.
>> Xorcom USB channel banks seems something quite interesting. Is there
>> anyone that could/would share experiences using that?
>> We need to replace an old PBX interfaced to 50 FXS and 8 BRI ISDN in
>> Italy.
>> My concern is about reliability of USB
>> Any success stories with it? Tips and tricks?
>>
>
>> Gilles wrote:
>> > On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez
>> >  wrote:
>> >
>> > >   Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports.
>> > >
>> >
>> > Thanks for the tip. It looks like the smallest option is 8 FXO ports:
>> >
>> > www.xorcom.com/telephony-interfaces/astribank-models.html
>> >
>> >
>        We use them a lot for high density analog lines and extensions.  The
> only thing to keep in mind is to always connect the units in a
> predetermined order to the USB ports so you do not mess up your
> configuration.  Apart from that they are really easy to use since the
> drivers are included in the standard dahdi distribution.
>
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001

I am sure that Tzafrir can pipe in here.  There is an method of
setting the ID of each astribank to keep them in order.  Ask Xorcom
for more info.

-- 
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Re: [asterisk-users] USB or Ethernet based FXO device ?

[Carlos Chavez]

On Wed, 2011-08-31 at 17:03 +0200, Marco Signorini wrote:
> Hi.
> 
> I was following this thread. We normally use Patton SmartNode SN4112
> series to interface to FXO ports. But I'm looking for something
> different for a future setup.
> Xorcom USB channel banks seems something quite interesting. Is there
> anyone that could/would share experiences using that? 
> We need to replace an old PBX interfaced to 50 FXS and 8 BRI ISDN in
> Italy.
> My concern is about reliability of USB
> Any success stories with it? Tips and tricks?
> 

> Gilles wrote: 
> > On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez
> >  wrote:
> >   
> > >   Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports.
> > > 
> > 
> > Thanks for the tip. It looks like the smallest option is 8 FXO ports:
> > 
> > www.xorcom.com/telephony-interfaces/astribank-models.html
> > 
> > 
We use them a lot for high density analog lines and extensions.  The
only thing to keep in mind is to always connect the units in a
predetermined order to the USB ports so you do not mess up your
configuration.  Apart from that they are really easy to use since the
drivers are included in the standard dahdi distribution.


-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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[asterisk-users] cli command show codecs

[RSCL Mumbai]

Hi,

Is there a CLI command which will tell me the codec used for active calls
and if transcoding is happening ?

Thx
Sans
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Re: [asterisk-users] Wanted a modified SIP message body

[Olle E. Johansson]

31 aug 2011 kl. 14:42 skrev Kevin P. Fleming:

> On 08/31/2011 02:46 AM, Jaime Lozano wrote:
>> Hello,
>> I agree with you, I'm not explaining the problem in a proper manner,
>> because of my lack of Asterisk knowings. I send the Wireshark captures.
>> 
>> 3com telephones take the timezone TZ:7200 from the 3Com PBX to show the
>> time right. But what if I want a 3Com telephone to work with Asterisk
>> PBX? Then the telephone time is wrong, 2 hours lower. It seems 3Com
>> telephones need the TZ:7200. 3Com telephones work with Asterisk and it
>> is great, but we would like to log the calls.
> 
> OK, so the first clarification is that you are talking about responses to 
> REGISTER requests specifically, not all responses to all requests. That's 
> good :-)
> 
> On to the meat of the issue... indeed, the '200 OK' response to a REGISTER 
> request does not normally have a message body; nothing in the SIP RFCs even 
> suggests that there would be one (although it's certainly allowed should the 
> registrar want to include it) or what would be present in it.
> 
> As has been previously replied here, there is no facility in Asterisk to 
> include a message body in a REGISTER request response, so providing one will 
> definitely require source code modifications. They wouldn't be terribly 
> difficult, but they would only be applicable to these particular phones, 
> which reduces the benefit of making the changes to the community at large.
> 
> With that said... it's certainly possible to do this, but it's going to take 
> some non-trivial code changes. The REGISTER handling code does not use any of 
> the methods that exist in chan_sip to add message body content to its 
> responses, it uses simpler methods that assume there won't be a message body.
> 
> In addition, this mechanism is really pretty broken anyway; the server would 
> have to know where each phone is physically located in order to be able to 
> provide the correct TZ value to it, and would have to be updated if the phone 
> is moved. Not an ideal situation.

The RFC states that a phone could use the Date: header in the response to set 
the local time in the device. It's always in GMT which makes it stupid to add a 
time zone any where. 

-1 for this implementation.

/O
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Re: [asterisk-users] USB or Ethernet based FXO device ?

[Cary Fitch]

We have run some tests on the Xorcom equipment, mostly the PRI port cards,
running up to 16 ports in a chassis.  They work.  I see no problem in Xorcom
as FXO ports.

 

We will be installing a lot of them as PRI ports soon..

 

CF

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco
Signorini
Sent: Wednesday, August 31, 2011 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] USB or Ethernet based FXO device ?

 

Hi.

I was following this thread. We normally use Patton SmartNode SN4112 series
to interface to FXO ports. But I'm looking for something different for a
future setup.
Xorcom USB channel banks seems something quite interesting. Is there anyone
that could/would share experiences using that? 
We need to replace an old PBX interfaced to 50 FXS and 8 BRI ISDN in Italy.
My concern is about reliability of USB
Any success stories with it? Tips and tricks?

Thank you and regards,
Marco Signorini.

  -- 

http://www.ingegnitech.com/images/logo.gif

INGEGNI Tech S.r.l.
site   http://www.ingegnitech.com
mail   i...@ingegnitech.com

  _  

 



Gilles wrote: 

On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez
   wrote:
  

  Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports.


 
Thanks for the tip. It looks like the smallest option is 8 FXO ports:
 
www.xorcom.com/telephony-interfaces/astribank-models.html
 
 
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Re: [asterisk-users] USB or Ethernet based FXO device ?

[Marco Signorini]

Hi.

I was following this thread. We normally use Patton SmartNode SN4112
series to interface to FXO ports. But I'm looking for something
different
for a future setup.
Xorcom USB channel banks seems something quite interesting. Is there
anyone that could/would share experiences using that? 
We need to replace an old PBX interfaced to 50 FXS and 8 BRI ISDN in
Italy.
My concern is about reliability of USB
Any success stories with it? Tips and tricks?

Thank you and regards,
Marco Signorini.

-- 



INGEGNI Tech S.r.l.
site http://www.ingegnitech.com
mail i...@ingegnitech.com






Gilles wrote:

  On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez
 wrote:
  
  
	Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports.

  
  
Thanks for the tip. It looks like the smallest option is 8 FXO ports:

www.xorcom.com/telephony-interfaces/astribank-models.html


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Re: [asterisk-users] T.38 passthru on 1.8.5

[Fabian Borot]

Hi Kevin, I created the issue on the https://issues.asterisk.org/jira web site, 
posted the description of the prob and submitted asterisk console logs [sip and 
udptl debug on] and a wireshark capture taken on the asterisk machine showing 
both legs with signaling and media.
PLease let me know what other thing you need you need me to provide.
Again we thank you in advance
fborot


 

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: RE: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 19:24:15 -0400







Txs a lot Kevin.
I had just created and account on https://issues.asterisk.org/jira
Let me know if this is the right place to post both the pcap capture and the 
sip logs. If not please help me out creating the account in the right place so 
that I can provide all the information you need.
The sip debug logs I can post here but I need to change the real IPs, which is 
easy to do because it will be a text file.
I appreciate your time and effort in helping us find the roout cause.
Fborot

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: RE: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 13:53:25 -0400




txs a lot for your explanation steve
so, it should work w/o spandsp fairly fine if we do not have a bad connection. 
I see that this version has a lot of fixes related to t.38
 but is the implementation already mature enough to guarantee a decent success 
rate with fax calls?


 

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: RE: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 13:15:19 -0400








will installing spandsp help with t.38 pass-through?


 

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: RE: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 11:42:41 -0400








both endpoints use public Ips, I just changed the real ones for the privates 
ones to protect our ips but made a mistake and left the dest as a pub and the 
orig as private, my bad.
but for the record, both are public IPs, there is no nat and iptables is off

 also, I see that the quintum sends a lot of these packages but asterisk sends 
only 1 or 2 to the other side.




 

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 09:44:15 -0400










 Hello
We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built 
by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 
21:31:22 UTC]

The call flow is:
quintum gateway --> asterisk --> Dialogic IMG 1010

the call starts as a voice call, the remote fax picks up and we hear the fax 
tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is 
passed back to the user side [quintum gateway] whcih reply with 200 OK with 
t.38 and the nothing else happens. After 20 secs of "inactivity" the quintum 
sends another Invite with voice only and then a BYE.

We do see that the quintum sends a lot of messages like this from the quintum's 
IP [192.168.1.18] but we do not see that asterisk sends the packages to the 
destination

UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, 
len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)

we have this settings on sip.conf
faxdetect = yes
t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations 
t38pt_udptl = yes;t38pt_udptl = yes,fec etc]

When we send the fax from the quintum to the Dialogic IMG the fax works 100% of 
the times.
I enabled fax set debug on and udptl set debug on but the console does not show 
almost anything but the udptl packets shown above.
What else should I do?Any ideas/help is greatly appreciated

txs a lot
fborot



  
   

Re: [asterisk-users] T.38 passthru on 1.8.5

[Kevin P. Fleming]

On 08/30/2011 06:24 PM, Fabian Borot wrote:

Txs a lot Kevin.
I had just created and account on https://issues.asterisk.org/jira
Let me know if this is the right place to post both the pcap capture and
the sip logs. If not please help me out creating the account in the
right place so that I can provide all the information you need.
The sip debug logs I can post here but I need to change the real IPs,
which is easy to do because it will be a text file.
I appreciate your time and effort in helping us find the roout cause.


Yes, that is the correct location.

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Dragging the dialup customers along, possible?

[Kevin P. Fleming]

On 08/29/2011 10:32 PM, C F wrote:

On Mon, Aug 29, 2011 at 4:32 PM, Cary Fitch  wrote:



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Monday, August 29, 2011 3:18 PM
To: t...@ewebforce.net; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Dragging the dialup customers along, possible?

 From what you are asking it appears that you are trying to run similar
to a fax (modulation and demodulation) over VoIP.
Try again, the fact that you succeeded twice was pure luck, and as far
as I understand that didn't even work out.
Switch back to TDM. Your dial up modems want that magic thing called
timing and no jitter that only TDM will give you.

===

This is more of a whimsical statement than a scientific one, but I would
think in today's world, there would be a real small box that would take in
IP and put out TDM with good timing with a moderate buffering window.
Obviously, the IP has to actually get to the box in a timely fashion, like
"today" , but a TDM circuit has to be "up" also.

A box that would take in IP data..., look for valid "ascii", and otherwise
put out TDM modem tones with no data content for "1 second" and then pick up
the data as it catches up.


So you want to develop the equivalent of T.38 for dial up?


It already exists; it's called V.150.

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Wanted a modified SIP message body

[Kevin P. Fleming]

On 08/31/2011 02:46 AM, Jaime Lozano wrote:

Hello,
I agree with you, I'm not explaining the problem in a proper manner,
because of my lack of Asterisk knowings. I send the Wireshark captures.

3com telephones take the timezone TZ:7200 from the 3Com PBX to show the
time right. But what if I want a 3Com telephone to work with Asterisk
PBX? Then the telephone time is wrong, 2 hours lower. It seems 3Com
telephones need the TZ:7200. 3Com telephones work with Asterisk and it
is great, but we would like to log the calls.


OK, so the first clarification is that you are talking about responses 
to REGISTER requests specifically, not all responses to all requests. 
That's good :-)


On to the meat of the issue... indeed, the '200 OK' response to a 
REGISTER request does not normally have a message body; nothing in the 
SIP RFCs even suggests that there would be one (although it's certainly 
allowed should the registrar want to include it) or what would be 
present in it.


As has been previously replied here, there is no facility in Asterisk to 
include a message body in a REGISTER request response, so providing one 
will definitely require source code modifications. They wouldn't be 
terribly difficult, but they would only be applicable to these 
particular phones, which reduces the benefit of making the changes to 
the community at large.


With that said... it's certainly possible to do this, but it's going to 
take some non-trivial code changes. The REGISTER handling code does not 
use any of the methods that exist in chan_sip to add message body 
content to its responses, it uses simpler methods that assume there 
won't be a message body.


In addition, this mechanism is really pretty broken anyway; the server 
would have to know where each phone is physically located in order to be 
able to provide the correct TZ value to it, and would have to be updated 
if the phone is moved. Not an ideal situation.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] how to play wav files to all members in konference

[virendra bhati]

Hi List,

Any hint ?

Is there any posibility to stream played file os any chanels to all members
?

On Sat, Aug 27, 2011 at 6:03 PM, virendra bhati  wrote:

> Hi List,
>
> How to play wav files to all konference members at a time. I want to play
> with the help of AMI connection.
>
> I have tested that we can play channel base file playing. But it will take
> too much time if users are more then 20
>
>
> -
> Thanks and regards
>
>  Virendra Bhati
> +91-9172341457
> Software Engineer
>
>


-- 



-
Thanks and regards

 Virendra Bhati
+91-9172341457
Software Engineer
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Re: [asterisk-users] Wanted a modified SIP message body

[Paul Hayes]

On 31/08/11 08:46, Jaime Lozano wrote:

Hello,
I agree with you, I'm not explaining the problem in a proper manner,
because of my lack of Asterisk knowings. I send the Wireshark captures.

3com telephones take the timezone TZ:7200 from the 3Com PBX to show the
time right. But what if I want a 3Com telephone to work with Asterisk
PBX? Then the telephone time is wrong, 2 hours lower. It seems 3Com
telephones need the TZ:7200. 3Com telephones work with Asterisk and it
is great, but we would like to log the calls.



Well I think your only options are to either modify the Asterisk source 
code to add this non-standard message body to your SIP packets or swap 
the phones for some that can use NTP which is how everyone one else sets 
the time on a network device.  These phones are obviously designed to 
only really ever be used with the 3COM PBX they were made for.


cheers,
Paul.

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Re: [asterisk-users] Wanted a modified SIP message body

[Jaime Lozano]

Hello,
I agree with you, I'm not explaining the problem in a proper manner, because
of my lack of Asterisk knowings. I send the Wireshark captures.

3com telephones take the timezone TZ:7200 from the 3Com PBX to show the time
right. But what if I want a 3Com telephone to work with Asterisk PBX? Then
the telephone time is wrong, 2 hours lower. It seems 3Com telephones need
the TZ:7200. 3Com telephones work with Asterisk and it is great, but we
would like to log the calls.

Ask me whatever you want.

Have a nice day

2011/8/30 Kevin P. Fleming 

> On 08/30/2011 07:36 AM, Jaime Lozano wrote:
>
>  I have been using wireshark to capture some traffic. I'm talking when
>> the PBX sends OK (200) connection accepted. 3Com PBX sends "TZ=7200\n"
>> (an much more things) in a SIP packet message body but Asterisk PBX
>> sends packets without message body, it only sends variables in the
>> message header. So I want Asterisk to send packets with a message body
>> and its proper content.
>>
>
> This is extremely confusing, to say the least. 'a message body' and 'its
> proper content' are ambiguous, especially since Asterisk already works with
> pretty much every SIP UA on the planet and none of them require the things
> you are asking for.
>
> Why don't you post an actual example of what you think Asterisk should be
> sending, and what it is actually sending, rather than trying to describe the
> differences (which is clearly not going well)?
>
> In general, though, you can't just put random content in a SIP request or
> response message body; the message body is usually of a defined type
> (application/sdp, for example), and has rules about what it can and cannot
> contain.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> 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
>

No. TimeSourceDestination   Protocol Length 
Info
 94 196.240917  10.100.190.3 10.100.0.244  SIP  1214   
Status: 200 OK(1 bindings)

+ Frame 94: 1214 bytes on wire (9712 bits), 1214 bytes captured (9712 bits)
+ Ethernet II, Src: Netscreen_ff:28:03 (00:10:db:ff:28:03), Dst: Nbx_32:2d:cb 
(00:e0:bb:32:2d:cb)
+ Internet Protocol Version 4, Src: 10.100.190.3 (10.100.190.3), Dst: 
10.100.0.244 (10.100.0.244)
+ User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
- Session Initiation Protocol
   - Status-Line: SIP/2.0 200 OK
Status-Code: 200
[Resent Packet: False]
[Request Frame: 92]
[Response Time (ms): 57]
   - Message Header
   + via: SIP/2.0/UDP 10.100.0.244:5060
   + from: 
   + to: ;tag=adf40a9c
 call-id: 92d91998-01d6-0672-13cb-00e0bb322dcb
   + cseq: 4 REGISTER
 date: Fri, 26 Aug 2011 07:58:37 GMT
   + contact: ;dt=546
 Expires: 2421
 user-agent: 3Com VCX 7210 IP CallProcessor/v7.1.210
 content-type: application/x-cw-user-profile
 content-length: 761
   - Message Body
   VER:1.0
   REG:y;3600
   TZ:7200
   DN:Comunicaciones 2
   DTF:dd MM HH:mm
   CIDS:n;889;890;;
   RA:1;1;8
   RD:1;1;8
   RO:1;1;8
   RCW:1;1;8
   DRNG:1560;3;6;8;0;
   
DR:4;4;5;0010;4;4;5;0088;5;5;5;01006;4;4;5;0012;4;4;5;0016;4;4;5;0060;5;5;5;01004;6;6;5;44;6;6;5;0118;4;4;5;2;11;11;5;9;6;6;5;33;10;10;5;080;4;32;5;000;10;10;5;09;4;4;5;0112;4;4;5;0061;10;10;5;090;4;4;5;15;4;4;5;5;4;4;5;6;4;4;5;7;4;4;5;9;5;5;5;8;4;4;5;0062;4;4;5;0080;4;4;5;0085;4;4;5;0091;4;4;5;0092;10;10;5;06;4;4;5;45;4;4;5;46;4;4;5;48;4;4;5;49;3;3;5;112;4;4;5;30;4;4;5;31;4;4;5;32;4;4;5;36;4;4;5;37;4;4;5;38;4;4;5;39;4;4;5;40;4;4;5;41;4;4;5;42;4;4;5;16;4;4;5;17;4;4;5;18;4;4;5;19;
   ACC:ES;es;10
   CP:2662
   CF:n;465;0615389973;n;467;;n;466;
   CD:n;446;n;440
   MB:Comunicaciones;1560;
   HG1:Comunicaciones;1560;y;
   LN:3
   QOS:n;0;0;0;0





No. TimeSourceDestination   Protocol Length 
Info
  7 3.31541710.100.190.7 10.100.0.244  SIP  516
Status: 200 OK(1 bindings)

+ Frame 7: 516 bytes on wire (4128 bits), 516 bytes captured (4128 bits)
+ Ethernet II, Src: Netscreen_ff:28:03 (00:10:db:ff:28:03), Dst: Nbx_32:2d:cb 
(00:e0:bb:32:2d:cb)
+ Internet Protocol Version 4, Src: 10.100.190.7 (10.100.

[asterisk-users] Update: Celebrating 10 years SER SIP Router in Vienna: 8th September 2011

[Klaus Darilion]

Update: the location is fixed now, it happens in the Unibräu in the 
Alten AKH Campus. We start at 7pm.


Please find details at
http://sip-router.org/10-years-ser-vienna/

If you plan to attend please send me an e-mail to make sure that we have 
enough seats.


regards
Klaus


On 16.08.2011 11:38, Klaus Darilion wrote:

Hi!

If you haven't noticed yet, SER (the mother of the SIP proxy projects
Openser, Kamailio, sip-router, opensips, ) is celebrating their 10th
year. There will be a main event happening in Berlin
(http://sip-router.org/10-years-ser/).

For those who can not travel to this event, there will be some smaller
events around the world.

In Vienna, this will happen on the 8th September at 7pm in the evening,
organized by Andreas Granig and myself. We do not plan to have a big
event with speaker, it is just an informal come together to meet with
other SER-interested[1] guys, having some techie talks, maybe getting
some business contacts and drinking some beers. Drinks will be sponsored
by http://www.ipcom.at/ and http://www.sipwise.com/.

The event will happen in the city center of Vienna. The location is not
fixed yet, it depends on how many people are going to attend. Thus,
please let us know if you going to attend (send us a short e-mail) and
spread this e-mail to interested people you know.

Cheers
Andreas and Klaus

[1] User's of all SER-flavors are welcome


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