Re: [asterisk-users] Asterisk Upgrade

[Tzafrir Cohen]

On Fri, Jan 16, 2009 at 01:24:16PM +, Gordon Henderson wrote:
> On Fri, 16 Jan 2009, Alex Balashov wrote:
> 
> > 1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk
> 
> I'd suggest not removing /etc/asterisk if that's the only source of your 
> config files... If you (re)generate them from elsewhere, it's probably OK.

/var/lib/asterisk/sounds may have some custom sound files .
/var/spool/asterisk/voicemail has voicemail and voicemail prompts.

> 
> and the important one, I'd have thought is
> 
>/usr/lib/asterisk/modules



-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] mini-PCI FXS card?

[Paul Chambers]

The EdgePBX FX08 has up to eight ports (using Digium/compatible 
modules), a couple of ethernet interfaces, and runs Astfin2 (Asterisk 
1.4.21 and uClinux 2.6.22 on a Blackfin DSP).

http://www.edgepbx.cn/shop/index.php?controller=product&product_id=6

or you might prefer its baby brother, the FX02, with up to two ports, 
and one ethernet interface:


http://www.edgepbx.cn/shop/index.php?controller=product&path=19&product_id=1

I recently purchased a FX08 for home use, and have been having a blast 
with it. I bought it with one FxO and one FxS module, then added some 
Digium modules I had from a TDM400.

-- Paul

Adam Moffett wrote:
> Is there any product that's a single port mini-PCI FXS card?
> I'm aware of the Openvox A400M 
> , but I really 
> only wanted one port.
>
> How about a single or dual port PCI or PCI express FXS card?
>
> Basically I wanted to build a small linux router with one or two phone 
> ports. 
>
> Alternatively, is there already a router or single board computer with 
> FXS ports that I could run linux/asterisk on?
>
> 
>
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Re: [asterisk-users] Portech MV-378 with Asterisk

[Pascal Bruno]

Sorry for bothering you, but I got it, I just had to put # in callnum!



On Sat, Jan 17, 2009 at 1:44 AM, Pascal Bruno  wrote:

> I want to dial out using the sim card.  What I did, I have used the SIP
> channel ex:
>
> Channel: SIP/thenum...@mv378
>
> It shows the called is being made in the dialplan, but the number I have
> entered does not dial, it just goes straight to the specified dialplan
> extensions.
>
> Then what I did, in the Lan to Mobile Table, I put * in url and the number
> I wanted to dial in call num, then the call was made to that number using
> the sim card properly.
>
> I was wondering if I cannot supply the number to be dialed using an
> asterisk call file, or do I have to put that number in the Lan to Mobile
> table.
>
> Any help would be appreciated.
>
> Thanks
>
>
>
>
>
> On Sat, Jan 17, 2009 at 12:39 AM, Pascal Bruno  wrote:
>
>> Marco,
>>
>> The configs work fine for me.  I can receive calls with no problem.  Now,
>> were you able to dial using the sim card?  I cant figure out how I can do it
>> since asterisk doesnt have a channel to place call through the portech
>> gateway.
>>
>>
>>
>>
>>
>> On Fri, Jan 16, 2009 at 12:04 PM, Pascal Bruno wrote:
>>
>>> Thank you!, I will try that in a few hours and let you know what happens.
>>>
>>>
>>>
>>> On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini 
>>> wrote:
>>>


 Pascal Bruno wrote:

 Thanks for your reply!

 Can you tell me what you have in your Portech configuration settings
 (Mobile to Lan Settings; Sip Proxy settings etc...)  My sip.conf file is
 pretty similar to yours but still cant register.



 On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini 
 wrote:

> Emmanuel Pascal Bruno wrote:
>
>  Has anyone been able to configure portech's mv-378 gateway with
> asterisk?
>
> I did the configuration as per the manual but it does not work.
>
> My server sees the portech gateway, but when the gateway is trying to
> register to my server it fails.  It says peer is not suppose to register.
>
> The gateway and the asterisk box are on two different location (two
> network, 2 differrent IP address).
>
> I would appreciate any kind of tutorial or advice on how to make it
> work.
>
> Thanks
>
> --
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> Hi,
> I've an installation working with Portech MV-370. I'm supposing it's
> quite similar to what you have. If it could be useful to you, this is my
> sip.conf configuration file.
>
> [GSMGtw1]
> type=friend
> context=from-gsm
> host=dynamic; we have a DHCP assigned address
> secret=reallyverysecret
> nat=no  ; there is not NAT between phone and
> Asterisk
> canreinvite=no
> dtmfmode=INFO
> insecure=invite ; required to overcome authentication
> problems in incoming calls
> call-limit=1   ; permit only 1 outgoing call at a
> time
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> qualify=500
>
> I remember that I've found a bug on the firmware that prevents to the
> unit to register correctly on my asterisk box unless I'm using the raw IP
> address instead of the name of the asterisk box. I remember something 
> wrong
> in cryptography chiper/dechiper based on realm... So, if you have 
> problems,
> let's try to specify the asterisk raw IP address in the Portech.
>
> Best regards,
> Marco Signorini.
>
>
>
 Hi,

 I don't know if the problem could be in the Mobile to Lan or Lan to
 Mobile settings because these  settings are related on how calls coming
 from/to mobile are routed.  I didn't use the Portech routing features at 
 all
 because I need a simple GSM gateway to/from the asterisk box.
 For this reason:
 1. The only rule I've on Mobile to Lan is CID=*; url=...@192.168.0.5where 
 "mob" is the extension I've generated in the asterisk box under the
 context where the Portech operates;
 2. The only rule I've on Lan to Mobile is URL=*; Call Num=#

 I think the most relevant parameters for your problem are under the
 "Service Domain" menu option (assuming that the firmware you have is 
 similar
 to what I've). On this menu I've compiled the 1st Realm (as I've only one
 account) like that:

 UserName: GSMGtw1
 RegisterName: GSMGtw1
 RegisterPassword: reallyverysecret
 Domain Server: 192.168.0.5
 Proxy Server: 192.168.0.5

 Pay attention that, having specified the Do

Re: [asterisk-users] Portech MV-378 with Asterisk

[Pascal Bruno]

I want to dial out using the sim card.  What I did, I have used the SIP
channel ex:

Channel: SIP/thenum...@mv378

It shows the called is being made in the dialplan, but the number I have
entered does not dial, it just goes straight to the specified dialplan
extensions.

Then what I did, in the Lan to Mobile Table, I put * in url and the number I
wanted to dial in call num, then the call was made to that number using the
sim card properly.

I was wondering if I cannot supply the number to be dialed using an asterisk
call file, or do I have to put that number in the Lan to Mobile table.

Any help would be appreciated.

Thanks





On Sat, Jan 17, 2009 at 12:39 AM, Pascal Bruno  wrote:

> Marco,
>
> The configs work fine for me.  I can receive calls with no problem.  Now,
> were you able to dial using the sim card?  I cant figure out how I can do it
> since asterisk doesnt have a channel to place call through the portech
> gateway.
>
>
>
>
>
> On Fri, Jan 16, 2009 at 12:04 PM, Pascal Bruno  wrote:
>
>> Thank you!, I will try that in a few hours and let you know what happens.
>>
>>
>>
>> On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini 
>> wrote:
>>
>>>
>>>
>>> Pascal Bruno wrote:
>>>
>>> Thanks for your reply!
>>>
>>> Can you tell me what you have in your Portech configuration settings
>>> (Mobile to Lan Settings; Sip Proxy settings etc...)  My sip.conf file is
>>> pretty similar to yours but still cant register.
>>>
>>>
>>>
>>> On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini 
>>> wrote:
>>>
 Emmanuel Pascal Bruno wrote:

  Has anyone been able to configure portech's mv-378 gateway with
 asterisk?

 I did the configuration as per the manual but it does not work.

 My server sees the portech gateway, but when the gateway is trying to
 register to my server it fails.  It says peer is not suppose to register.

 The gateway and the asterisk box are on two different location (two
 network, 2 differrent IP address).

 I would appreciate any kind of tutorial or advice on how to make it
 work.

 Thanks

 --

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 Hi,
 I've an installation working with Portech MV-370. I'm supposing it's
 quite similar to what you have. If it could be useful to you, this is my
 sip.conf configuration file.

 [GSMGtw1]
 type=friend
 context=from-gsm
 host=dynamic; we have a DHCP assigned address
 secret=reallyverysecret
 nat=no  ; there is not NAT between phone and
 Asterisk
 canreinvite=no
 dtmfmode=INFO
 insecure=invite ; required to overcome authentication
 problems in incoming calls
 call-limit=1   ; permit only 1 outgoing call at a
 time
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm
 qualify=500

 I remember that I've found a bug on the firmware that prevents to the
 unit to register correctly on my asterisk box unless I'm using the raw IP
 address instead of the name of the asterisk box. I remember something wrong
 in cryptography chiper/dechiper based on realm... So, if you have problems,
 let's try to specify the asterisk raw IP address in the Portech.

 Best regards,
 Marco Signorini.



>>> Hi,
>>>
>>> I don't know if the problem could be in the Mobile to Lan or Lan to
>>> Mobile settings because these  settings are related on how calls coming
>>> from/to mobile are routed.  I didn't use the Portech routing features at all
>>> because I need a simple GSM gateway to/from the asterisk box.
>>> For this reason:
>>> 1. The only rule I've on Mobile to Lan is CID=*; url=...@192.168.0.5where 
>>> "mob" is the extension I've generated in the asterisk box under the
>>> context where the Portech operates;
>>> 2. The only rule I've on Lan to Mobile is URL=*; Call Num=#
>>>
>>> I think the most relevant parameters for your problem are under the
>>> "Service Domain" menu option (assuming that the firmware you have is similar
>>> to what I've). On this menu I've compiled the 1st Realm (as I've only one
>>> account) like that:
>>>
>>> UserName: GSMGtw1
>>> RegisterName: GSMGtw1
>>> RegisterPassword: reallyverysecret
>>> Domain Server: 192.168.0.5
>>> Proxy Server: 192.168.0.5
>>>
>>> Pay attention that, having specified the Domain Server with the raw IP
>>> address, asterisk needs to be able to authenticate peers associated to that.
>>> For this reason I've set:
>>>
>>> domain=192.168.0.5
>>>
>>> on sip.conf [general] section (remember to issue a sip reload from
>>> asterisk cli).
>>>
>>> Hope this helps!
>>>
>>>
>>> Best regards.
>>> Mar

Re: [asterisk-users] Portech MV-378 with Asterisk

[Pascal Bruno]

Marco,

The configs work fine for me.  I can receive calls with no problem.  Now,
were you able to dial using the sim card?  I cant figure out how I can do it
since asterisk doesnt have a channel to place call through the portech
gateway.




On Fri, Jan 16, 2009 at 12:04 PM, Pascal Bruno  wrote:

> Thank you!, I will try that in a few hours and let you know what happens.
>
>
>
> On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini wrote:
>
>>
>>
>> Pascal Bruno wrote:
>>
>> Thanks for your reply!
>>
>> Can you tell me what you have in your Portech configuration settings
>> (Mobile to Lan Settings; Sip Proxy settings etc...)  My sip.conf file is
>> pretty similar to yours but still cant register.
>>
>>
>>
>> On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini wrote:
>>
>>> Emmanuel Pascal Bruno wrote:
>>>
>>>  Has anyone been able to configure portech's mv-378 gateway with
>>> asterisk?
>>>
>>> I did the configuration as per the manual but it does not work.
>>>
>>> My server sees the portech gateway, but when the gateway is trying to
>>> register to my server it fails.  It says peer is not suppose to register.
>>>
>>> The gateway and the asterisk box are on two different location (two
>>> network, 2 differrent IP address).
>>>
>>> I would appreciate any kind of tutorial or advice on how to make it work.
>>>
>>> Thanks
>>>
>>> --
>>>
>>> ___
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>> Hi,
>>> I've an installation working with Portech MV-370. I'm supposing it's
>>> quite similar to what you have. If it could be useful to you, this is my
>>> sip.conf configuration file.
>>>
>>> [GSMGtw1]
>>> type=friend
>>> context=from-gsm
>>> host=dynamic; we have a DHCP assigned address
>>> secret=reallyverysecret
>>> nat=no  ; there is not NAT between phone and
>>> Asterisk
>>> canreinvite=no
>>> dtmfmode=INFO
>>> insecure=invite ; required to overcome authentication
>>> problems in incoming calls
>>> call-limit=1   ; permit only 1 outgoing call at a
>>> time
>>> disallow=all
>>> allow=ulaw
>>> allow=alaw
>>> allow=gsm
>>> qualify=500
>>>
>>> I remember that I've found a bug on the firmware that prevents to the
>>> unit to register correctly on my asterisk box unless I'm using the raw IP
>>> address instead of the name of the asterisk box. I remember something wrong
>>> in cryptography chiper/dechiper based on realm... So, if you have problems,
>>> let's try to specify the asterisk raw IP address in the Portech.
>>>
>>> Best regards,
>>> Marco Signorini.
>>>
>>>
>>>
>> Hi,
>>
>> I don't know if the problem could be in the Mobile to Lan or Lan to Mobile
>> settings because these  settings are related on how calls coming from/to
>> mobile are routed.  I didn't use the Portech routing features at all because
>> I need a simple GSM gateway to/from the asterisk box.
>> For this reason:
>> 1. The only rule I've on Mobile to Lan is CID=*; url=...@192.168.0.5where 
>> "mob" is the extension I've generated in the asterisk box under the
>> context where the Portech operates;
>> 2. The only rule I've on Lan to Mobile is URL=*; Call Num=#
>>
>> I think the most relevant parameters for your problem are under the
>> "Service Domain" menu option (assuming that the firmware you have is similar
>> to what I've). On this menu I've compiled the 1st Realm (as I've only one
>> account) like that:
>>
>> UserName: GSMGtw1
>> RegisterName: GSMGtw1
>> RegisterPassword: reallyverysecret
>> Domain Server: 192.168.0.5
>> Proxy Server: 192.168.0.5
>>
>> Pay attention that, having specified the Domain Server with the raw IP
>> address, asterisk needs to be able to authenticate peers associated to that.
>> For this reason I've set:
>>
>> domain=192.168.0.5
>>
>> on sip.conf [general] section (remember to issue a sip reload from
>> asterisk cli).
>>
>> Hope this helps!
>>
>>
>> Best regards.
>> Marco Signorini
>>
>>
>>
>> 
>> Marco Signorini
>> INGEGNI Tech S.r.l.
>> http://www.ingegnitech.com
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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[asterisk-users] UpdateConfig : Appending line fails

[Jose P. Espinal]

Thank you very much, Tilghman, 
I did not respond before because for some reason the IP of the list server was 
blacklisted on spamhaus.org and I was not getting the messages

see:


2009-01-16 22:47:49 H=(lists.digium.com) [216.207.245.1] 
F= rejected RCPT 
: "JunkMail rejected - (lists.digium.com) [216.207.245.1] 
is in an RBL, see http://www.spamhaus.org/query/bl?ip=216.207.245.1";



About UpdateConfig syntax, how did  you find out the correct way of sending 
various sets of parameters?
I was looking in google, the ATFOT v2 Book, and nothing showed up.



Thanks again, regards.





--
Jose P. Espinal

 
 

>/
/>/
/>/ I'm I missing something?
/
Perhaps you actually meant to do the following command:

Action: UpdateConfig
SrcFilename: voicemail2.conf
DstFilename: voicemail2.conf
Action-00: Append
Cat-00: default
Var-00: 127
Value-00: >, Jason Bourne97, jason25 at noCia.gov.do 

Action-01: Append
Cat-01: default
Var-01: 125
Value-01: >5, Jason Bourne76, jason12 at noCia.gov.do 

Action-02: Append
Cat-02: default
Var-02: 122
Value-02: >, Jason Bourne74, jason49 at noCia.gov.do 

Action-03: Append
Cat-03: default
Var-03: 128
Value-03: >, Jason Bourne48, jason45 at noCia.gov.do 

Action-04: Append
Cat-04: default
Var-04: 126
Value-04: >5, Jason Bourne18, jason64 at noCia.gov.do 

ActionID: 495446608

-- 
Tilghman


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Re: [asterisk-users] CDR Rewrite -- Questions to the users

[Grey Man]

On Sat, Jan 17, 2009 at 3:08 AM, Steve Murphy  wrote:
> Greyman--
>
> I've been thinking of Benny Amorsen's comments on Simple CDRs...
>
>
>> This is tricky... I need to create these CDR's for the billing system:
>>
>> src: A  start: e1  ans: e2  end: e6   dst: B  disp: ANSW
>> src: A  start: e4  ans: e5  end: e6   dst: B  disp: ANSW
>>
>> If I do the same substitution again, I get this:
>>
>> A->B start: e1 ans: e2 end: e4. Whoops, end time is wrong.
>> A->C start: e1 ans: e5 end: e4. Whoops, both start and end times are
>> wrong.
>>
>> CDR2 needs to find e1 so it can replace start, while CDR3 shouldn't
>> have anything replaced. I can't think of a query which will do this
>> correctly.
>
> Do you have any ideas that might make this work for him?
> I guess simple CDR's won't work for everyone, but... if they
> could work in this case
>

>From what I can gather the problem Benny has is getting the src field
correct for a transfer. In the traditional Telco World the src (or A
Number) field tends to be both the callerid of the customer and an
identifier that ties the CDR to the customer for billing purposes.

With Asterisk and a lot of other modern day softswitches there's
usually a field called accountcode or similar which can be used to tie
a CDR to a customer. The src field is then only fulfilling one role
which is to hold the callerid that was set or recieved for the call.

The trick with transfers is to forget about the src field for billing
purposes and make sure the accountcode for the call is set in
accordance with the business rules. For example if two customers A and
B are talking to each other and A blind transfers B to a billable
destination Z then who pays for the call from B to Z? There is no
right answer but as far as the CDRs are concerned it's irrelvant as
long as each call is recorded and the accountcode can be set within
the dialplan both choices can be accomodated.

With the simple CDR approach it could end up that there are multiple
CDRs for a customer for the one call since they could be charged for
both ends of a transfer. I don't see that as an issue.

Regards,

Greyman.

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Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

[sean darcy]

Tilghman Lesher wrote:
> On Friday 16 January 2009 17:43:21 sean darcy wrote:
>> Danny Nicholas wrote:
>>> Why not do a "zap restart" instead of restarting asterisk?  You could
>>> write an AGI to do the ZR when the condition occurred and lines where
>>> empty.
>> Yes, a cron job to restart zaptel would cut off any call then existing.
>>
>> But how would I test for it? I can imagine:
>>
>> exten=>s,n,ExecIf(, System(service dahdi restart))
>>
>> It's the  I can't imagine. How do you test if dahdi is
>> acting up?
> 
> Not a service restart, but a "dahdi restart".  You can't restart the dahdi
> service without first stopping Asterisk, anyway.
> 
> if [ `../asterisk-trunk/contrib/scripts/astcli "core show channels" | wc -l` 
> = 
> 3 ]; then asterisk -rx 'dahdi restart'; fi
> 

Wow. I'll try that tomorrow. Put it as the cmd right after answer(), 
right? Or maybe, h,1 ?

Well anyway, at least I'll be able to receive calls over pstn with dahdi.

Thanks.

sean


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Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

[Tilghman Lesher]

On Friday 16 January 2009 17:43:21 sean darcy wrote:
> Danny Nicholas wrote:
> > Why not do a "zap restart" instead of restarting asterisk?  You could
> > write an AGI to do the ZR when the condition occurred and lines where
> > empty.
>
> Yes, a cron job to restart zaptel would cut off any call then existing.
>
> But how would I test for it? I can imagine:
>
> exten=>s,n,ExecIf(, System(service dahdi restart))
>
> It's the  I can't imagine. How do you test if dahdi is
> acting up?

Not a service restart, but a "dahdi restart".  You can't restart the dahdi
service without first stopping Asterisk, anyway.

if [ `../asterisk-trunk/contrib/scripts/astcli "core show channels" | wc -l` = 
3 ]; then asterisk -rx 'dahdi restart'; fi

-- 
Tilghman

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Re: [asterisk-users] UpdateConfig : Appending line fails

[Tilghman Lesher]

On Friday 16 January 2009 16:47:40 Jose P. Espinal wrote:
> Can someone please point me out why would a stream like the following
> only write ONE line (the first) on the given file?
>
> Action: login
> Username: test
> Secret: 123456
>
> Action: UpdateConfig
> SrcFilename: voicemail2.conf
> DstFilename: voicemail2.conf
> Action-00: Append
> Cat-00: default
> Var-00: 127
> Value-00: >, Jason Bourne97, jaso...@nocia.gov.do
> ActionID: 1256187957

This is a single valid command, and so it works.

> Action: UpdateConfig
> SrcFilename: voicemail2.conf
> DstFilename: voicemail2.conf
> Action-01: Append
> Cat-01: default
> Var-01: 125
> Value-01: >5, Jason Bourne76, jaso...@nocia.gov.do
> ActionID: 1607673137

This command did not specify an Action-00.

> Action: UpdateConfig
> SrcFilename: voicemail2.conf
> DstFilename: voicemail2.conf
> Action-02: Append
> Cat-02: default
> Var-02: 122
> Value-02: >, Jason Bourne74, jaso...@nocia.gov.do
> ActionID: 165797792

This command did not specify an Action-00.

> Action: UpdateConfig
> SrcFilename: voicemail2.conf
> DstFilename: voicemail2.conf
> Action-03: Append
> Cat-03: default
> Var-03: 128
> Value-03: >, Jason Bourne48, jaso...@nocia.gov.do
> ActionID: 1743636529

This command did not specify an Action-00.

> Action: UpdateConfig
> SrcFilename: voicemail2.conf
> DstFilename: voicemail2.conf
> Action-04: Append
> Cat-04: default
> Var-04: 126
> Value-04: >5, Jason Bourne18, jaso...@nocia.gov.do
> ActionID: 495446608

This command did not specify an Action-00.

> Action: Logoff
> Username: test
>
>
> I'm I missing something?

Perhaps you actually meant to do the following command:

Action: UpdateConfig
SrcFilename: voicemail2.conf
DstFilename: voicemail2.conf
Action-00: Append
Cat-00: default
Var-00: 127
Value-00: >, Jason Bourne97, jaso...@nocia.gov.do
Action-01: Append
Cat-01: default
Var-01: 125
Value-01: >5, Jason Bourne76, jaso...@nocia.gov.do
Action-02: Append
Cat-02: default
Var-02: 122
Value-02: >, Jason Bourne74, jaso...@nocia.gov.do
Action-03: Append
Cat-03: default
Var-03: 128
Value-03: >, Jason Bourne48, jaso...@nocia.gov.do
Action-04: Append
Cat-04: default
Var-04: 126
Value-04: >5, Jason Bourne18, jaso...@nocia.gov.do
ActionID: 495446608

-- 
Tilghman

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Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

[sean darcy]

Danny Nicholas wrote:
> Why not do a "zap restart" instead of restarting asterisk?  You could write
> an AGI to do the ZR when the condition occurred and lines where empty.
> 
Yes, a cron job to restart zaptel would cut off any call then existing.

But how would I test for it? I can imagine:

exten=>s,n,ExecIf(, System(service dahdi restart))

It's the  I can't imagine. How do you test if dahdi is 
acting up?

sean


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Re: [asterisk-users] CDR problems -- two call legs create only one CDR. Using ForkCDR() not even working.

[Grey Man]

On Fri, Jan 16, 2009 at 6:12 PM, Örn Arnarson  wrote:
> Am I missing something? Any ideas appreciated.

No you are not missing anything. The Asterisk CDR implementation has a
number of issues and one CDR per bridge is one of them. There is
currently a re-design discussion going on on the list at the moment.
If you're really interested you can read the new design document at
http://svn.digium.com/svn/asterisk/team/murf/RFCs/CDRfix2.rfc.txt.

Regards,

Greyman.

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[asterisk-users] UpdateConfig : Appending line fails

[Jose P. Espinal]

Hello list,

Can someone please point me out why would a stream like the following 
only write ONE line (the first) on the given file?

Action: login
Username: test
Secret: 123456

Action: UpdateConfig
SrcFilename: voicemail2.conf
DstFilename: voicemail2.conf
Action-00: Append
Cat-00: default
Var-00: 127
Value-00: >, Jason Bourne97, jaso...@nocia.gov.do
ActionID: 1256187957

Action: UpdateConfig
SrcFilename: voicemail2.conf
DstFilename: voicemail2.conf
Action-01: Append
Cat-01: default
Var-01: 125
Value-01: >5, Jason Bourne76, jaso...@nocia.gov.do
ActionID: 1607673137

Action: UpdateConfig
SrcFilename: voicemail2.conf
DstFilename: voicemail2.conf
Action-02: Append
Cat-02: default
Var-02: 122
Value-02: >, Jason Bourne74, jaso...@nocia.gov.do
ActionID: 165797792

Action: UpdateConfig
SrcFilename: voicemail2.conf
DstFilename: voicemail2.conf
Action-03: Append
Cat-03: default
Var-03: 128
Value-03: >, Jason Bourne48, jaso...@nocia.gov.do
ActionID: 1743636529

Action: UpdateConfig
SrcFilename: voicemail2.conf
DstFilename: voicemail2.conf
Action-04: Append
Cat-04: default
Var-04: 126
Value-04: >5, Jason Bourne18, jaso...@nocia.gov.do
ActionID: 495446608

Action: Logoff
Username: test


I'm I missing something?

PS. (I receive 'success' for all of the updating attempts)


Thanks in advice,


--
Jose P. Espinal
http://blog.slackware-es.com

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Re: [asterisk-users] Snom 300 vs Grandstream gxp

[Hans Witvliet]

On Fri, 2009-01-16 at 15:52 +, Julian Lyndon-Smith wrote:
> Can anyone who has used both comment on the pros and cons ? Need to buy 
> about 30 of these, for a small company with limited IT support.
> 

For evaluation a got a couple of different phones.

gxp2000, cheap, but it works, some people (not me) have lots of problems
with them. 4-sip-users per phone. no VLAN support.

GXP-budget, even cheaper, but works, just one single sip-user per phone.

snom320, bit more expensive, also works great, easy to upgrade.
Latest version even got v6-support. all versions have VLAN support,
6-sip-users per phone. Easy to configure with web-interface.

Siemens optipoint 410, something to avoid! (even it it were less
expensive) If power/lan are plugged in wrong order, you get an instant
brick, as the firmware is wiped and no possibility to get it back.

I mention VLAN support, in case you don't have separate lan's it might
be worthwhile if a lot of data passes through the rest of the lan


hw

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Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

[Steve Totaro]

Both are hacks.  Neither are good hacks.

On Fri, Jan 16, 2009 at 5:04 PM, Danny Nicholas  wrote:
> So "Zap restart" is a worse hack than restarting *?  What DAHDI does that
> Zaptel doesn't -
> 1. Makes Digium lawyers happy
> 2. Gives Developers headaches
> 3. Makes Coffee and soft drink manufacturers happy since we need so much
> caffine.
> 4. Make priests happy since we have to go to confession about it.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
> Sent: Friday, January 16, 2009 3:54 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] pstn hangs up: MWI no message waiting ??
>
> Sounds like an awful hack.  What does DAHDI do that Zaptel does not?
> Sounds more like a post for the bugs list
>
> On Fri, Jan 16, 2009 at 4:48 PM, Danny Nicholas  wrote:
>> Why not do a "zap restart" instead of restarting asterisk?  You could
> write
>> an AGI to do the ZR when the condition occurred and lines where empty.
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
>> Sent: Friday, January 16, 2009 3:39 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] pstn hangs up: MWI no message waiting ??
>>
>> On Fri, Jan 16, 2009 at 4:10 PM, sean darcy  wrote:
>>> Doug Bailey wrote:
 - "sean darcy"  wrote:

> pstn incoming on a TDM400P, sometimes i* won't answer, going into
> a loop like this:
>
>   -- Starting simple switch on 'DAHDI/4-1'
> [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
>
> 18 (Ring Begin)...
> [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
> 2
> (Ring/Answered)...
> [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI:
> Channel 4 no message waiting!

 This message occurs when the pstn sends a FSK spill indicating the
 message waiting status of the FXO port in question.   This may encoded
 in the caller ID indicator or may be contained in its own message spill.
 This is output as a NOTICE logging message.

 Regards,
 Doug Bailey

>>>
>>> I'm not sure I understand all that, but why does asterisk hang up? It
>>> means I can't receive any calls on that pstn line. AFAICS, only
>>> restarting asterisk allows calls to be received.
>>>
>>> A cron job to restart every 5 minutes?
>>>
>>> sean
>>>
>>>
>>
>> I hear Zaptel is pretty stable..
>>
>> --
>> Thanks,
>> Steve Totaro
>> +18887771888 (Toll Free)
>> +12409381212 (Cell)
>> +12024369784 (Skype)
>>
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
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-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

[Danny Nicholas]

So "Zap restart" is a worse hack than restarting *?  What DAHDI does that
Zaptel doesn't -
1. Makes Digium lawyers happy
2. Gives Developers headaches
3. Makes Coffee and soft drink manufacturers happy since we need so much
caffine.
4. Make priests happy since we have to go to confession about it.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Friday, January 16, 2009 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

Sounds like an awful hack.  What does DAHDI do that Zaptel does not?
Sounds more like a post for the bugs list

On Fri, Jan 16, 2009 at 4:48 PM, Danny Nicholas  wrote:
> Why not do a "zap restart" instead of restarting asterisk?  You could
write
> an AGI to do the ZR when the condition occurred and lines where empty.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
> Sent: Friday, January 16, 2009 3:39 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] pstn hangs up: MWI no message waiting ??
>
> On Fri, Jan 16, 2009 at 4:10 PM, sean darcy  wrote:
>> Doug Bailey wrote:
>>> - "sean darcy"  wrote:
>>>
 pstn incoming on a TDM400P, sometimes i* won't answer, going into
 a loop like this:

   -- Starting simple switch on 'DAHDI/4-1'
 [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event

 18 (Ring Begin)...
 [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
 2
 (Ring/Answered)...
 [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI:
 Channel 4 no message waiting!
>>>
>>> This message occurs when the pstn sends a FSK spill indicating the
>>> message waiting status of the FXO port in question.   This may encoded
>>> in the caller ID indicator or may be contained in its own message spill.
>>> This is output as a NOTICE logging message.
>>>
>>> Regards,
>>> Doug Bailey
>>>
>>
>> I'm not sure I understand all that, but why does asterisk hang up? It
>> means I can't receive any calls on that pstn line. AFAICS, only
>> restarting asterisk allows calls to be received.
>>
>> A cron job to restart every 5 minutes?
>>
>> sean
>>
>>
>
> I hear Zaptel is pretty stable..
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

[Steve Totaro]

Sounds like an awful hack.  What does DAHDI do that Zaptel does not?
Sounds more like a post for the bugs list

On Fri, Jan 16, 2009 at 4:48 PM, Danny Nicholas  wrote:
> Why not do a "zap restart" instead of restarting asterisk?  You could write
> an AGI to do the ZR when the condition occurred and lines where empty.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
> Sent: Friday, January 16, 2009 3:39 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] pstn hangs up: MWI no message waiting ??
>
> On Fri, Jan 16, 2009 at 4:10 PM, sean darcy  wrote:
>> Doug Bailey wrote:
>>> - "sean darcy"  wrote:
>>>
 pstn incoming on a TDM400P, sometimes i* won't answer, going into
 a loop like this:

   -- Starting simple switch on 'DAHDI/4-1'
 [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event

 18 (Ring Begin)...
 [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
 2
 (Ring/Answered)...
 [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI:
 Channel 4 no message waiting!
>>>
>>> This message occurs when the pstn sends a FSK spill indicating the
>>> message waiting status of the FXO port in question.   This may encoded
>>> in the caller ID indicator or may be contained in its own message spill.
>>> This is output as a NOTICE logging message.
>>>
>>> Regards,
>>> Doug Bailey
>>>
>>
>> I'm not sure I understand all that, but why does asterisk hang up? It
>> means I can't receive any calls on that pstn line. AFAICS, only
>> restarting asterisk allows calls to be received.
>>
>> A cron job to restart every 5 minutes?
>>
>> sean
>>
>>
>
> I hear Zaptel is pretty stable..
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

[Danny Nicholas]

Why not do a "zap restart" instead of restarting asterisk?  You could write
an AGI to do the ZR when the condition occurred and lines where empty.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Friday, January 16, 2009 3:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

On Fri, Jan 16, 2009 at 4:10 PM, sean darcy  wrote:
> Doug Bailey wrote:
>> - "sean darcy"  wrote:
>>
>>> pstn incoming on a TDM400P, sometimes i* won't answer, going into
>>> a loop like this:
>>>
>>>   -- Starting simple switch on 'DAHDI/4-1'
>>> [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
>>>
>>> 18 (Ring Begin)...
>>> [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
>>> 2
>>> (Ring/Answered)...
>>> [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI:
>>> Channel 4 no message waiting!
>>
>> This message occurs when the pstn sends a FSK spill indicating the
>> message waiting status of the FXO port in question.   This may encoded
>> in the caller ID indicator or may be contained in its own message spill.
>> This is output as a NOTICE logging message.
>>
>> Regards,
>> Doug Bailey
>>
>
> I'm not sure I understand all that, but why does asterisk hang up? It
> means I can't receive any calls on that pstn line. AFAICS, only
> restarting asterisk allows calls to be received.
>
> A cron job to restart every 5 minutes?
>
> sean
>
>
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> To UNSUBSCRIBE or update options visit:
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I hear Zaptel is pretty stable..

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

[Danny Nicholas]

Yes, BUT  ..  not 100% and discontinued in 1.4.22 on ...

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Friday, January 16, 2009 3:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

On Fri, Jan 16, 2009 at 4:10 PM, sean darcy  wrote:
> Doug Bailey wrote:
>> - "sean darcy"  wrote:
>>
>>> pstn incoming on a TDM400P, sometimes i* won't answer, going into
>>> a loop like this:
>>>
>>>   -- Starting simple switch on 'DAHDI/4-1'
>>> [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
>>>
>>> 18 (Ring Begin)...
>>> [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
>>> 2
>>> (Ring/Answered)...
>>> [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI:
>>> Channel 4 no message waiting!
>>
>> This message occurs when the pstn sends a FSK spill indicating the
>> message waiting status of the FXO port in question.   This may encoded
>> in the caller ID indicator or may be contained in its own message spill.
>> This is output as a NOTICE logging message.
>>
>> Regards,
>> Doug Bailey
>>
>
> I'm not sure I understand all that, but why does asterisk hang up? It
> means I can't receive any calls on that pstn line. AFAICS, only
> restarting asterisk allows calls to be received.
>
> A cron job to restart every 5 minutes?
>
> sean
>
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
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I hear Zaptel is pretty stable..

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

[Steve Totaro]

On Fri, Jan 16, 2009 at 4:10 PM, sean darcy  wrote:
> Doug Bailey wrote:
>> - "sean darcy"  wrote:
>>
>>> pstn incoming on a TDM400P, sometimes i* won't answer, going into
>>> a loop like this:
>>>
>>>   -- Starting simple switch on 'DAHDI/4-1'
>>> [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
>>>
>>> 18 (Ring Begin)...
>>> [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
>>> 2
>>> (Ring/Answered)...
>>> [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI:
>>> Channel 4 no message waiting!
>>
>> This message occurs when the pstn sends a FSK spill indicating the
>> message waiting status of the FXO port in question.   This may encoded
>> in the caller ID indicator or may be contained in its own message spill.
>> This is output as a NOTICE logging message.
>>
>> Regards,
>> Doug Bailey
>>
>
> I'm not sure I understand all that, but why does asterisk hang up? It
> means I can't receive any calls on that pstn line. AFAICS, only
> restarting asterisk allows calls to be received.
>
> A cron job to restart every 5 minutes?
>
> sean
>
>
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I hear Zaptel is pretty stable..

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

[sean darcy]

Doug Bailey wrote:
> - "sean darcy"  wrote:
> 
>> pstn incoming on a TDM400P, sometimes i* won't answer, going into
>> a loop like this:
>>
>>   -- Starting simple switch on 'DAHDI/4-1'
>> [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
>>
>> 18 (Ring Begin)...
>> [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
>> 2 
>> (Ring/Answered)...
>> [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: 
>> Channel 4 no message waiting!
> 
> This message occurs when the pstn sends a FSK spill indicating the 
> message waiting status of the FXO port in question.   This may encoded 
> in the caller ID indicator or may be contained in its own message spill. 
> This is output as a NOTICE logging message. 
> 
> Regards, 
> Doug Bailey 
> 

I'm not sure I understand all that, but why does asterisk hang up? It 
means I can't receive any calls on that pstn line. AFAICS, only 
restarting asterisk allows calls to be received.

A cron job to restart every 5 minutes?

sean


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Re: [asterisk-users] ATA gateway with 2 ethernet interfaces

[Jeff LaCoursiere]

Agg, I felt bad about being pedantic.  How about splitting the load and 
reducing the single point of failure?  Instead of one big ATA how about a 
number of smaller ones (two port) split between your switches?

j

On Fri, 16 Jan 2009, Jeff LaCoursiere wrote:

>
>
> I would be more worried about the ATA gateway failing than the switch, as
> you have found yourself.  How about two gateways and two phones on
> everyone's desk :)
>
> j
>
> On Fri, 16 Jan 2009, Adam Moffett wrote:
>
>> I don't know of any ATA like that except the grandstream.
>>
>> The service provider grade way to do this would probably be a Cisco (or
>> similar) with a T1 interface and a channel bank to break the T1 into 24
>> FXS ports.
>>
>>
>>
>>> Hi,
>>>
>>> I'm looking for an 8+ FXS ATA gateway (at least 8 ports but preferably at 
>>> most 24 ports) with 2 ethernet interfaces for network/switch redundancy.
>>>
>>> So far I've only found the Grandstream GXW4008.
>>>
>>> I've searched similar brands such as Linksys and higher-end brands such as 
>>> Quintum, but they all seem to have just one NIC. So, if the switch the ATA 
>>> is connected to fails then I'm out of business (at least until I replace 
>>> the switch but that's usually too long for a busy system).
>>>
>>> The GXW4008 device is very useful for this scenario. It has 2 RJ45 ports 
>>> (called WAN and LAN) and I've set them up in two local subnets. Not only 
>>> does the ATA keep working without human intervention if one of the switches 
>>> goes down but if both switches are up it can load balance between the two 
>>> (simply by using DNS SRV with the same weights).
>>>
>>> Unfortunately, Grandstream in general doesn't seem to be very reliable 
>>> although the latest GXW4008 firmware has proven to be quite stable in my 
>>> case (previous releases were buggy).
>>>
>>> So I'm looking for alternatives to the GXW4008, even if it has to cost me 
>>> more money. Does anyone know of an 8+ FXS ATA brand/model with 2 ethernet 
>>> interfaces?
>>>
>>> Thanks in advance.
>>>
>>> Vieri
>>>
>>>
>>>
>>>
>>>
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>>>
>>
>>
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Re: [asterisk-users] Asterisk - Trixbox

[Adrià Vidal]

On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett  wrote:
> My provider migrated from an old EOL softswitch to Trixbox.
>
> I have a number (8159093011) on a different server on a different network.
> It appears as though the incoming calls are trying to authenticate against
> that number, which isn't present on the box.  Could someone help me decode
> this debugging output?  I was calling 8159911010.  My server is
> 208.100.1.33.  Theirs is 208.1.87.235.  I solved the s@ problem on the other
> server by adding insecure settings, but that didn't seem to solve it on this
> one.
>
> http://pastebin.com/f5151341f
>
>
> -
> Mike Hammett
> Intelligent Computing Solutions
> http://www.ics-il.com



I think you need something inside [DID-incoming] like for example...


exten => s,1,NoOP(-incoming call---)
exten => s,n,Playback(wellcome)


#
Looking for s in DID-incoming (domain 208.100.1.33)
#
Reliably Transmitting (no NAT) to 208.1.87.235:5060:
#
SIP/2.0 404 Not Found


-- 
--
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adriavi...@gmail.com
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Re: [asterisk-users] ATA gateway with 2 ethernet interfaces

[Jeff LaCoursiere]

I would be more worried about the ATA gateway failing than the switch, as 
you have found yourself.  How about two gateways and two phones on 
everyone's desk :)

j

On Fri, 16 Jan 2009, Adam Moffett wrote:

> I don't know of any ATA like that except the grandstream.
>
> The service provider grade way to do this would probably be a Cisco (or
> similar) with a T1 interface and a channel bank to break the T1 into 24
> FXS ports.
>
>
>
>> Hi,
>>
>> I'm looking for an 8+ FXS ATA gateway (at least 8 ports but preferably at 
>> most 24 ports) with 2 ethernet interfaces for network/switch redundancy.
>>
>> So far I've only found the Grandstream GXW4008.
>>
>> I've searched similar brands such as Linksys and higher-end brands such as 
>> Quintum, but they all seem to have just one NIC. So, if the switch the ATA 
>> is connected to fails then I'm out of business (at least until I replace the 
>> switch but that's usually too long for a busy system).
>>
>> The GXW4008 device is very useful for this scenario. It has 2 RJ45 ports 
>> (called WAN and LAN) and I've set them up in two local subnets. Not only 
>> does the ATA keep working without human intervention if one of the switches 
>> goes down but if both switches are up it can load balance between the two 
>> (simply by using DNS SRV with the same weights).
>>
>> Unfortunately, Grandstream in general doesn't seem to be very reliable 
>> although the latest GXW4008 firmware has proven to be quite stable in my 
>> case (previous releases were buggy).
>>
>> So I'm looking for alternatives to the GXW4008, even if it has to cost me 
>> more money. Does anyone know of an 8+ FXS ATA brand/model with 2 ethernet 
>> interfaces?
>>
>> Thanks in advance.
>>
>> Vieri
>>
>>
>>
>>
>>
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Re: [asterisk-users] ATA gateway with 2 ethernet interfaces

[Adam Moffett]

I don't know of any ATA like that except the grandstream.

The service provider grade way to do this would probably be a Cisco (or 
similar) with a T1 interface and a channel bank to break the T1 into 24 
FXS ports.



> Hi,
>
> I'm looking for an 8+ FXS ATA gateway (at least 8 ports but preferably at 
> most 24 ports) with 2 ethernet interfaces for network/switch redundancy.
>
> So far I've only found the Grandstream GXW4008.
>
> I've searched similar brands such as Linksys and higher-end brands such as 
> Quintum, but they all seem to have just one NIC. So, if the switch the ATA is 
> connected to fails then I'm out of business (at least until I replace the 
> switch but that's usually too long for a busy system).
>
> The GXW4008 device is very useful for this scenario. It has 2 RJ45 ports 
> (called WAN and LAN) and I've set them up in two local subnets. Not only does 
> the ATA keep working without human intervention if one of the switches goes 
> down but if both switches are up it can load balance between the two (simply 
> by using DNS SRV with the same weights).
>
> Unfortunately, Grandstream in general doesn't seem to be very reliable 
> although the latest GXW4008 firmware has proven to be quite stable in my case 
> (previous releases were buggy).
>
> So I'm looking for alternatives to the GXW4008, even if it has to cost me 
> more money. Does anyone know of an 8+ FXS ATA brand/model with 2 ethernet 
> interfaces?
>
> Thanks in advance.
>
> Vieri
>
>
>
>   
>
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>   


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[asterisk-users] Asterisk 1.4.23-rc4 Now Available

[Asterisk Development Team]

The Asterisk.org development team has published Asterisk 1.4.23-rc4.
This release candidate is available for download from
http://downloads.digium.com/.

A number of critical issues have been resolved since the last release
candidate for 1.4.23.  We hope to have this be the final release
candidate.  If all goes well, 1.4.23 will be released early next week.

For a full list of changes, see the ChangeLog:

http://svn.digium.com/svn-view/asterisk/tags/1.4.23-rc4/ChangeLog?view=markup

Testing of this release candidate would be very much appreciated.  One
area that has received a lot of changes is the handling of call parking,
so that is an area that could use some special attention in testing.
Please report any issues found in testing to http://bugs.digium.com/.

Thank you for your support of Asterisk!

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[asterisk-users] Crickets. Yes, crickets.

[John Todd]

How many times have you been on a conference call and some other  
participant puts their line on hold, leading to their hold music  
making conversation impossible for the rest of the group?

This scenario happens to me all the time and it drives me NUTS.  I  
prefer no hold music, because I am often on conference calls, and I  
also usually hate listening to music on the phone when I usually have  
my own music playing in the background as well.  And most hold music  
is really bad.  However, there is a valid reason for hold music and  
that is to let the caller know via an audio cue that they have not  
been disconnected.

So I heard about some really great hold audio the other day.  It was  
the sound of crickets chirping quietly, every 5-8 seconds, seemingly  
randomly.  It was actually just a 2 or 3 minute recording of a few  
different cricket sounds, recorded at low gain.  It was unobtrusive  
enough to not interrupt speakers on the call, yet if it was for a  
single listener it was enough audio to be obvious that the other  
speaker was still "on hold".

I have made some of the sounds in the past (tt-monkeys.gsm comes to  
mind) but at the moment I am pretty much tapped out for time and even  
trivial things like making sound files are difficult, though I can  
type messages to asterisk-users well enough, as I sit ironically on  
hold.

Does anyone want to take up the task?  While as usual I can't  
guarantee inclusion of any resulting soundfile into Asterisk, it would  
certainly get _my_ vote.

JT

---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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[asterisk-users] ATA gateway with 2 ethernet interfaces

[Vieri]

Hi,

I'm looking for an 8+ FXS ATA gateway (at least 8 ports but preferably at most 
24 ports) with 2 ethernet interfaces for network/switch redundancy.

So far I've only found the Grandstream GXW4008.

I've searched similar brands such as Linksys and higher-end brands such as 
Quintum, but they all seem to have just one NIC. So, if the switch the ATA is 
connected to fails then I'm out of business (at least until I replace the 
switch but that's usually too long for a busy system).

The GXW4008 device is very useful for this scenario. It has 2 RJ45 ports 
(called WAN and LAN) and I've set them up in two local subnets. Not only does 
the ATA keep working without human intervention if one of the switches goes 
down but if both switches are up it can load balance between the two (simply by 
using DNS SRV with the same weights).

Unfortunately, Grandstream in general doesn't seem to be very reliable although 
the latest GXW4008 firmware has proven to be quite stable in my case (previous 
releases were buggy).

So I'm looking for alternatives to the GXW4008, even if it has to cost me more 
money. Does anyone know of an 8+ FXS ATA brand/model with 2 ethernet interfaces?

Thanks in advance.

Vieri



  

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[asterisk-users] about hardware

[David fire]

hi
i want to setup an asterisk in my home i always worked whit digium hardware
but have any one try the chines cards or the openbox or sangoma?
is just for home 2 FXS and 2 FXO i need an aceptable audio and the pc will
do only asterisk and is a dual core 1G RAM
the cheaper chines cards will be ok?
thanks


-- 
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
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Re: [asterisk-users] mini-PCI FXS card?

[Adam Moffett]

Thanks Philipp,

This and everything else I see out there is a bit more than I need :)

I'm sure a single or dual port analog FXS card is not something most 
people want though, otherwise somebody would be selling it.

Thanks anyway though.


>
> I'd recommend Sangoma's new B700 FlexBRI hybrid card (4 BRI ports,
> 2 FXS/FXO)
> http://www.sangoma.com/products_and_solutions/hardware/digital_analog_hybrids/flex_bri.html
> or the B600 (4 FXO, 1 FXS)
> http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/b600.html
> although that might be a bit mor than you need.
>
>
>Philipp Kempgen
>
>   


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Re: [asterisk-users] mini-PCI FXS card?

[Philipp Kempgen]

Adam Moffett schrieb:
> How about a single or dual port PCI or PCI express FXS card?
> 
> Basically I wanted to build a small linux router with one or two phone 
> ports. 

I'd recommend Sangoma's new B700 FlexBRI hybrid card (4 BRI ports,
2 FXS/FXO)
http://www.sangoma.com/products_and_solutions/hardware/digital_analog_hybrids/flex_bri.html
or the B600 (4 FXO, 1 FXS)
http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/b600.html
although that might be a bit mor than you need.


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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[asterisk-users] mini-PCI FXS card?

[Adam Moffett]

Is there any product that's a single port mini-PCI FXS card?
I'm aware of the Openvox A400M 
, but I really only 
wanted one port.


How about a single or dual port PCI or PCI express FXS card?

Basically I wanted to build a small linux router with one or two phone 
ports. 

Alternatively, is there already a router or single board computer with 
FXS ports that I could run linux/asterisk on?



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Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

[Doug Bailey]

- "sean darcy"  wrote:

> pstn incoming on a TDM400P, sometimes i* won't answer, going into
> a loop like this:
> 
>   -- Starting simple switch on 'DAHDI/4-1'
> [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
> 
> 18 (Ring Begin)...
> [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
> 2 
> (Ring/Answered)...
> [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: 
> Channel 4 no message waiting!

This message occurs when the pstn sends a FSK spill indicating the 
message waiting status of the FXO port in question.   This may encoded 
in the caller ID indicator or may be contained in its own message spill. 
This is output as a NOTICE logging message. 

Regards, 
Doug Bailey 

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Re: [asterisk-users] How to transfer a call from one AsteriskServerto another

[Paul]

Yes, this is the first method I tried.  The transfer only works if it is
done before a media path is set up to the first box (not answered by the
IVR).  If it is answered then transferred, I get a 500 internal server error
back from the ITSP and the call dies.  I never see anything hit the second
box.
 
 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Friday, January 16, 2009 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to transfer a call from one
AsteriskServerto another



I guess you already tried this? 

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

Thanks

l.





2009/1/16 Paul 


I do have it functioning with Dial().   I was looking for a way to
completely move the call from the first box though.  When using Dial() media
moves, but the call is still tied to the first box.  In looking at captures
when the call is ended, the first box invites out to the ITSP again, then
after receiving a 200ok sends a bye.
 
Also while testing, once the call was up on the second box, I stopped
Asterisk on the first box which kills the call.
 
 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Friday, January 16, 2009 12:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to transfer a call from one AsteriskServer
to another



Why don't you simply Dial() the call to a separate box keeping Asterisk out
of the audio path?

l.



2009/1/16 Paul 



Can anyone tell me how I can completely move an established call off of one
Asterisk server to another?
 
In our case we have a server with our IVR.  Depending upon digits entered,
the call can be transferred to any of our other servers depending where the
extension or queue reside.
We would like to completely move the call off of the first box so we don't
tie up resources on it.
 
In our lab we are testing with 1.4.22.1
 
Our provider which delivers inbound calls to us uses a Sonus gateway.   So
far, testing has shown that if we transfer the inbound call prior to any
media playback, it works.  But, if the IVR plays media, then it is failing,
with a 500 internal server error being returned.
 
Thanks for any help
 
 
 

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-- 
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Re: [asterisk-users] Voicemail message is dialtone

[Steve Johnson]

Here's also an example snip from the debug log:

[07:42:20] -- Executing [...@mainmenu:15] Dial("Zap/1-1",
"SIP/105|18|tKk") in new stack
[07:42:20] -- SIP/105-08571180 is ringing
[07:42:39] -- Nobody picked up in 18000 ms
[07:42:39] -- Executing [...@mainmenu:16] Answer("Zap/1-1", "") in new stack
[07:42:39] -- Executing [...@mainmenu:17] Wait("Zap/1-1", "1") in new stack
[07:42:40] -- Executing [...@mainmenu:18] Playback("Zap/1-1",
"silence/1") in new stack
[07:42:40] --  Playing 'silence/1' (language 'en')
[07:42:41] -- Executing [...@mainmenu:19] BackGround("Zap/1-1",
"please-leave-a-message") in new stack
[07:42:41] --  Playing 'please-leave-a-message' (language 'en')
[07:42:47] -- Executing [...@mainmenu:20] VoiceMail("Zap/1-1", "105|s")
in new stack
[07:42:48] --  Playing 'beep' (language 'en')
[07:42:48] -- Recording the message
[07:42:48] -- x=0, open writing:
/var/spool/asterisk/voicemail/default/100/tmp/UNCGT6 format: wav49,
0x8570eb8
[07:42:48] -- x=1, open writing:
/var/spool/asterisk/voicemail/default/100/tmp/UNCGT6 format: gsm,
0x8227430
[07:42:48] -- x=2, open writing:
/var/spool/asterisk/voicemail/default/100/tmp/UNCGT6 format: wav,
0x8259b10
[07:46:42] -- Recording automatically stopped after a silence of 10 seconds
[07:46:42] --  Playing 'auth-thankyou' (language 'en')
[07:46:43] == Auto fallthrough, channel 'Zap/1-1' status is 'NOANSWER'
[07:46:43] -- Hungup 'Zap/1-1'


On Fri, Jan 16, 2009 at 12:07 PM, Steve Johnson  wrote:
> Hello all,
>
> I have one Asterisk 1.4.21 system connected to a North American POTS
> line.  Normally hangup detection works fine, and Asterisk hangs up
> properly if you are talking to a caller and they hang up; but
> occasionally a call comes in (typically from a US telemarketer) where
> the caller hangs up just as voicemail recording is starting, and you
> get a voicemail recording of dialtone (then congestion and off-hook
> warning tones) for almost 4 minutes before asterisk gives up the line.
>
> zapata.conf [channels] options are:
>
> language=en
> context=default
> rxwink=300
> usecallerid=yes
> hidecallerid=no
> cidsignalling=bell
> callwaiting=no
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=no
> transfer=yes
> canpark=yes
> cancallforward=no
> callreturn=no
> musiconhold=default
> echocancel=yes
> echocancelwhenbridged=yes
> immediate=no
> faxdetect=no
> relaxdtmf=yes
> hanguponpolarityswitch=yes
> progzone=us
> signalling=fxs_ks
> channel => 1
>
>
> Any suggestions for voicemail detecting/rejecting messages when there
> is only dialtone on the other end?
>
> Thanks!
>
> S.
>

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[asterisk-users] CDR problems -- two call legs create only one CDR. Using ForkCDR() not even working.

[Örn Arnarson]

Hello,

When I bridge an incoming and outgoing call (attempting to simulate
call-forwarding) I'm only getting one CDR -- that of the outgoing call.

A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone
on PSTN) and bridges the call.
The only CDR created is from B to C. I have even tried using Answer() and
ForkCDR() to get two CDRs, but to no avail.
I am starting to wonder if there's a bug in the CDR generation in general,
because I set up an extension to do only that:

exten => 5822558,1,Answer()
exten => 5822558,n,ForkCDR()
exten => 5822558,n,Playback(tt-monkeys)
exten => 5822558,n,Hangup()

This is even given as an example on how to generate two CDRs from one call
on this website:
http://asterisk.name/asterisk/0596009623/asterisk-app-b-79.html

I have been able to create two CDRs with the use of the Local/n channel, but
the CDR is messy if I do so because I am required by law to change the
caller-id for the outgoing call to that of the PBX, so both call legs seem
to be originating from the Asterisk.

Am I missing something? Any ideas appreciated.

Best regards,
Örn
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Re: [asterisk-users] Dialing from E1/T1

[David fire]

exten => _X,1,Dial(DAHDI/g1/${EXTEN})
or
exten => _9,1,Dial(DAHDI/g1/${EXTEN:1})

the first whan put X to match the amount of digits.

the second one dial if you put a nine before. and many X as digits

David

2009/1/16 Gabriel Ortiz Lour 

> Hi,
>
>   A have an asterisk connected to a legacy PBX trought an E1 and to the
> PSTN trought another E1. When the legacy user dial to the PSTN the call pass
> trought Asterisk.
>
>   All works OK, the only problem is the delay on the Asterisk server when
> it receives the digits from the 1st E1 link. It will only make the call when
> the digit timeout expires.
>
>   Is there a way to make something like dialplan/context/exten match, so it
> will dial as soon as there a match?
>
> Thanks
> Gabriel Ortiz
>
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Re: [asterisk-users] How to transfer a call from one AsteriskServer to another

[Lenz Emilitri]

I guess you already tried this?

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

Thanks

l.



2009/1/16 Paul 

>  I do have it functioning with Dial().   I was looking for a way to
> completely move the call from the first box though.  When using Dial() media
> moves, but the call is still tied to the first box.  In looking at captures
> when the call is ended, the first box invites out to the ITSP again, then
> after receiving a 200ok sends a bye.
>
> Also while testing, once the call was up on the second box, I stopped
> Asterisk on the first box which kills the call.
>
>
>
>  --
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri
> *Sent:* Friday, January 16, 2009 12:17 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] How to transfer a call from one
> AsteriskServer to another
>
> Why don't you simply Dial() the call to a separate box keeping Asterisk out
> of the audio path?
>
> l.
>
> 2009/1/16 Paul 
>
>>  Can anyone tell me how I can completely move an established call off of
>> one Asterisk server to another?
>>
>> In our case we have a server with our IVR.  Depending upon digits entered,
>> the call can be transferred to any of our other servers depending where the
>> extension or queue reside.
>> We would like to completely move the call off of the first box so we don't
>> tie up resources on it.
>>
>> In our lab we are testing with 1.4.22.1
>>
>> Our provider which delivers inbound calls to us uses a Sonus gateway.   So
>> far, testing has shown that if we transfer the inbound call prior to any
>> media playback, it works.  But, if the IVR plays media, then it is failing,
>> with a 500 internal server error being returned.
>>
>> Thanks for any help
>>
>>
>>
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
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>
>
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[asterisk-users] Voicemail message is dialtone

[Steve Johnson]

Hello all,

I have one Asterisk 1.4.21 system connected to a North American POTS
line.  Normally hangup detection works fine, and Asterisk hangs up
properly if you are talking to a caller and they hang up; but
occasionally a call comes in (typically from a US telemarketer) where
the caller hangs up just as voicemail recording is starting, and you
get a voicemail recording of dialtone (then congestion and off-hook
warning tones) for almost 4 minutes before asterisk gives up the line.

zapata.conf [channels] options are:

language=en
context=default
rxwink=300
usecallerid=yes
hidecallerid=no
cidsignalling=bell
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=no
transfer=yes
canpark=yes
cancallforward=no
callreturn=no
musiconhold=default
echocancel=yes
echocancelwhenbridged=yes
immediate=no
faxdetect=no
relaxdtmf=yes
hanguponpolarityswitch=yes
progzone=us
signalling=fxs_ks
channel => 1


Any suggestions for voicemail detecting/rejecting messages when there
is only dialtone on the other end?

Thanks!

S.

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Re: [asterisk-users] How to hangup a call manually...

[Danny Nicholas]

If you’re using the GUI it will hang it up.  Otherwise “sip reload” might do
it.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grygoriy
Dobrovolskyy
Sent: Friday, January 16, 2009 12:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to hangup a call manually...

 

try to know the whole string ?
core show channels

2009/1/16 Carlos Chavez 

   I have this call:

SIP/protel-525512047 default  90445528885371  1 Ringing
AppDial  (Outgoing Line)   90445528885371  264:24:2
(None)

   I cannot use the soft hangup commando from the CLI because I do not
know the whole SIP channel string.  What other command can I use to
terminate this call or to find the complete channel string to put into
soft hangup?

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

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Re: [asterisk-users] How to hangup a call manually...

[D Tucny]

2009/1/17 Carlos Chavez 

>I have this call:
>
> SIP/protel-525512047 default  90445528885371  1 Ringing
> AppDial  (Outgoing Line)   90445528885371  264:24:2
> (None)
>
>I cannot use the soft hangup commando from the CLI because I do not
> know the whole SIP channel string.  What other command can I use to
> terminate this call or to find the complete channel string to put into
> soft hangup?
>

It looks like you have the entire channel string there, but, if not... The
CLI support tab completion, so

soft hangup SIP/protel-

will either fully autocomplete, or, where there are multiple matches, give
you a list of matching channels...

d
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Re: [asterisk-users] How to hangup a call manually...

[Grygoriy Dobrovolskyy]

try to know the whole string ?
core show channels

2009/1/16 Carlos Chavez 

>I have this call:
>
> SIP/protel-525512047 default  90445528885371  1 Ringing
> AppDial  (Outgoing Line)   90445528885371  264:24:2
> (None)
>
>I cannot use the soft hangup commando from the CLI because I do not
> know the whole SIP channel string.  What other command can I use to
> terminate this call or to find the complete channel string to put into
> soft hangup?
>
> --
> 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] How to hangup a call manually...

[Carlos Chavez]

I have this call:

SIP/protel-525512047 default  90445528885371  1 Ringing
AppDial  (Outgoing Line)   90445528885371  264:24:2
(None)

I cannot use the soft hangup commando from the CLI because I do not
know the whole SIP channel string.  What other command can I use to
terminate this call or to find the complete channel string to put into
soft hangup?

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


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Re: [asterisk-users] How to transfer a call from one AsteriskServer to another

[Paul]

I do have it functioning with Dial().   I was looking for a way to
completely move the call from the first box though.  When using Dial() media
moves, but the call is still tied to the first box.  In looking at captures
when the call is ended, the first box invites out to the ITSP again, then
after receiving a 200ok sends a bye.
 
Also while testing, once the call was up on the second box, I stopped
Asterisk on the first box which kills the call.
 
 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Friday, January 16, 2009 12:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to transfer a call from one AsteriskServer
to another



Why don't you simply Dial() the call to a separate box keeping Asterisk out
of the audio path?

l.



2009/1/16 Paul 



Can anyone tell me how I can completely move an established call off of one
Asterisk server to another?
 
In our case we have a server with our IVR.  Depending upon digits entered,
the call can be transferred to any of our other servers depending where the
extension or queue reside.
We would like to completely move the call off of the first box so we don't
tie up resources on it.
 
In our lab we are testing with 1.4.22.1
 
Our provider which delivers inbound calls to us uses a Sonus gateway.   So
far, testing has shown that if we transfer the inbound call prior to any
media playback, it works.  But, if the IVR plays media, then it is failing,
with a 500 internal server error being returned.
 
Thanks for any help
 
 
 

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[asterisk-users] Dialing from E1/T1

[Gabriel Ortiz Lour]

Hi,

  A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN
trought another E1. When the legacy user dial to the PSTN the call pass
trought Asterisk.

  All works OK, the only problem is the delay on the Asterisk server when it
receives the digits from the 1st E1 link. It will only make the call when
the digit timeout expires.

  Is there a way to make something like dialplan/context/exten match, so it
will dial as soon as there a match?

Thanks
Gabriel Ortiz
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Re: [asterisk-users] Snom 300 vs Grandstream gxp

[Daniel Hazelbaker]

On Jan 16, 2009, at 7:52 AM, Julian Lyndon-Smith wrote:

> Can anyone who has used both comment on the pros and cons ? Need to  
> buy
> about 30 of these, for a small company with limited IT support.

We recently deployed 85 phones to our office.  We tested the  
Grandstream GXP2000, GXP2020, Linksys SPA941, Snom 300 & 320, and a  
Polycom 430 (I think that was the series).  As an IT department we  
expected everybody to prefer the Grandstream because it is simple to  
use.  We figured everybody would have the Snom because it is complex  
to use (though super easy on the IT side to administer).  We had the  
opposite result.  Everybody hated the Grandstream because they sounded  
bad, felt clunky, were difficult to do simple things on (like park a  
call, can't do it with one button).  Nobody really cared for either  
the Linksys or Polycom.  They were just too limited.

We ended up rolling out a mixture of the Snom 300 and 320s and  
couldn't be happier (We looked at the 360, but it really doesn't offer  
anything except a bigger display, which isn't really utilized).  With  
a simple MySQL database and a few PHP scripts all we had to do was  
type the MAC address of the phone into the MySQL database (with the  
login information) and then plug the phones in.  No setup on the  
phone.  Phone automatically upgrades the firmware to whatever version  
we currently use, gets its settings from the server, etc.  If a phone  
has trouble (out of the 85 we had 2 that were a bit finicky and got  
replaced), we go into the database and change the MAC address and then  
plug in the new phone.  Again, no setup.

If you go Snom I would be happy to share these scripts, I just haven't  
gotten around to building up a nice package and posting them.  If your  
choices are either Snom or Grandstream, I would so go Snom.  I spent 2  
days trying to configure the GXP's to do the few simple things we  
wanted and couldn't pull it off (call parking, BLF & one-touch dial  
[does not fully work], etc).  I spent 30 minutes on the Snom an had it  
perfectly configured.

>> Julian


Daniel

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Re: [asterisk-users] Portech MV-378 with Asterisk

[Pascal Bruno]

Thank you!, I will try that in a few hours and let you know what happens.



On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini wrote:

>
>
> Pascal Bruno wrote:
>
> Thanks for your reply!
>
> Can you tell me what you have in your Portech configuration settings
> (Mobile to Lan Settings; Sip Proxy settings etc...)  My sip.conf file is
> pretty similar to yours but still cant register.
>
>
>
> On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini wrote:
>
>> Emmanuel Pascal Bruno wrote:
>>
>>  Has anyone been able to configure portech's mv-378 gateway with
>> asterisk?
>>
>> I did the configuration as per the manual but it does not work.
>>
>> My server sees the portech gateway, but when the gateway is trying to
>> register to my server it fails.  It says peer is not suppose to register.
>>
>> The gateway and the asterisk box are on two different location (two
>> network, 2 differrent IP address).
>>
>> I would appreciate any kind of tutorial or advice on how to make it work.
>>
>> Thanks
>>
>> --
>>
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>>
>> Hi,
>> I've an installation working with Portech MV-370. I'm supposing it's quite
>> similar to what you have. If it could be useful to you, this is my sip.conf
>> configuration file.
>>
>> [GSMGtw1]
>> type=friend
>> context=from-gsm
>> host=dynamic; we have a DHCP assigned address
>> secret=reallyverysecret
>> nat=no  ; there is not NAT between phone and
>> Asterisk
>> canreinvite=no
>> dtmfmode=INFO
>> insecure=invite ; required to overcome authentication
>> problems in incoming calls
>> call-limit=1   ; permit only 1 outgoing call at a time
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> allow=gsm
>> qualify=500
>>
>> I remember that I've found a bug on the firmware that prevents to the unit
>> to register correctly on my asterisk box unless I'm using the raw IP address
>> instead of the name of the asterisk box. I remember something wrong in
>> cryptography chiper/dechiper based on realm... So, if you have problems,
>> let's try to specify the asterisk raw IP address in the Portech.
>>
>> Best regards,
>> Marco Signorini.
>>
>>
>>
> Hi,
>
> I don't know if the problem could be in the Mobile to Lan or Lan to Mobile
> settings because these  settings are related on how calls coming from/to
> mobile are routed.  I didn't use the Portech routing features at all because
> I need a simple GSM gateway to/from the asterisk box.
> For this reason:
> 1. The only rule I've on Mobile to Lan is CID=*; url=...@192.168.0.5 where
> "mob" is the extension I've generated in the asterisk box under the context
> where the Portech operates;
> 2. The only rule I've on Lan to Mobile is URL=*; Call Num=#
>
> I think the most relevant parameters for your problem are under the
> "Service Domain" menu option (assuming that the firmware you have is similar
> to what I've). On this menu I've compiled the 1st Realm (as I've only one
> account) like that:
>
> UserName: GSMGtw1
> RegisterName: GSMGtw1
> RegisterPassword: reallyverysecret
> Domain Server: 192.168.0.5
> Proxy Server: 192.168.0.5
>
> Pay attention that, having specified the Domain Server with the raw IP
> address, asterisk needs to be able to authenticate peers associated to that.
> For this reason I've set:
>
> domain=192.168.0.5
>
> on sip.conf [general] section (remember to issue a sip reload from asterisk
> cli).
>
> Hope this helps!
>
>
> Best regards.
> Marco Signorini
>
>
>
> 
> Marco Signorini
> INGEGNI Tech S.r.l.
> http://www.ingegnitech.com
>
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Re: [asterisk-users] Remote RTP

[Mark Michelson]

Gabriel Ortiz Lour wrote:
> Hi all,
> 
>   Suposing that 2 SIP phone register at a remote (internet) asterisk, 
> what is the best way, if any, to make the RTP traffic go phone to phone, 
> whithout using the internet conection (asterisk)?
> 
> Thanks,
> Gabriel
> 

By default, Asterisk will attempt to offload the media from the server so that 
it may flow directly between the phones.

There are several factors which may prevent this, though. For instance, if 
Asterisk is recording the call or needs to listen for DTMF in order to activate 
a specific feature, then Asterisk has to have the RTP flow through it.

Mark Michelson

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Re: [asterisk-users] Remote RTP

[Gabriel Ortiz Lour]

They will be in the same LAN, probably behind NAT.

Being in the same LAN helps something?

2009/1/16 Jerry Jones 

>
> On Jan 16, 2009, at 10:38 AM, Gabriel Ortiz Lour wrote:
>
> > Hi all,
> >
> >   Suposing that 2 SIP phone register at a remote (internet)
> > asterisk, what is the best way, if any, to make the RTP traffic go
> > phone to phone, whithout using the internet conection (asterisk)?They
>
> Allow reinvite? Assuming both are not behind NAT.
>
>
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Re: [asterisk-users] Remote RTP

[Jerry Jones]

On Jan 16, 2009, at 10:38 AM, Gabriel Ortiz Lour wrote:

> Hi all,
>
>   Suposing that 2 SIP phone register at a remote (internet)  
> asterisk, what is the best way, if any, to make the RTP traffic go  
> phone to phone, whithout using the internet conection (asterisk)?

Allow reinvite? Assuming both are not behind NAT.


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Re: [asterisk-users] Remote RTP

[Alex Balashov]

canreinvite=yes.

Gabriel Ortiz Lour wrote:

> Hi all,
> 
>   Suposing that 2 SIP phone register at a remote (internet) asterisk, 
> what is the best way, if any, to make the RTP traffic go phone to phone, 
> whithout using the internet conection (asterisk)?
> 
> Thanks,
> Gabriel
> 
> 
> 
> 
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-- 
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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[asterisk-users] Remote RTP

[Gabriel Ortiz Lour]

Hi all,

  Suposing that 2 SIP phone register at a remote (internet) asterisk, what
is the best way, if any, to make the RTP traffic go phone to phone, whithout
using the internet conection (asterisk)?

Thanks,
Gabriel
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Re: [asterisk-users] Snom 300 vs Grandstream gxp

[Gordon Henderson]

On Fri, 16 Jan 2009, Julian Lyndon-Smith wrote:

> Can anyone who has used both comment on the pros and cons ? Need to buy
> about 30 of these, for a small company with limited IT support.

You get more phone for your money with the Grandstream, but ...

You'll hear a lot of people here who've had bad results with them in the 
past, so will be baised against them, and there were early issues with 
software quality - and still some issues with newer software on holder 
handsets, however ...

I've deployed a lot of Grandstreams - 100's. Mostly GXP2000s but one place 
(a small school on a tight budget) went for BT200's with a single GXP2000 
for the reception and they're happy. If budget was that tight, I'd look at 
the GXP280's though.

Audio quality in the Snoms is probably better. Build quality is a little 
better too - the handsets are slightly heavier.

I find call transfers easier on the Grandstreams - also the display (on 
the GXP range) is much bigger than that Snom 300's. That's nice if you're 
putting up number->names.

The web interface on both is easy enough if you're not auto provisioning. 
(I don't auto provision the grandstreams as such, but use a utility called 
'gsutil' on them) However, while it's very comprehensive, I think there's 
too much information there!

Well - that's my input... (mostly on Grandstreams though!)

Gordon

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Re: [asterisk-users] Snom 300 vs Grandstream gxp

[Robert Broyles]

Hi,

I've never used Snom phones, but have used the Grandstreams.
I think you will find that they just feel 'cheap.'  We had a half dozen 
of them, and the functionality is there, and they work great. But they 
just feel rough and cheap when using them.
If you are planning on using different headsets with them, you are fine. 
But if you are planning on using the factory headsets, you might find 
that the headset has rough edges, etc.

Call me 'crazy'!  We're using Linksys SPA-942/941's and couldn't be 
happier. The 941 model is a dollar or so more than the GXP, but don't 
have dual Ethernet. 942's do, for an extra $20.

Regards,
Robert Broyles


Julian Lyndon-Smith wrote:
> Can anyone who has used both comment on the pros and cons ? Need to buy 
> about 30 of these, for a small company with limited IT support.
>
> Julian
>
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Re: [asterisk-users] Portech MV-378 with Asterisk

[Marco Signorini]

Pascal Bruno wrote:
> Thanks for your reply!
>  
> Can you tell me what you have in your Portech configuration settings
> (Mobile to Lan Settings; Sip Proxy settings etc...)  My sip.conf file
> is pretty similar to yours but still cant register.
>
>
>  
> On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini  > wrote:
>
> Emmanuel Pascal Bruno wrote:
>> Has anyone been able to configure portech's mv-378 gateway with
>> asterisk?
>>  
>> I did the configuration as per the manual but it does not work.
>>  
>> My server sees the portech gateway, but when the gateway is
>> trying to register to my server it fails.  It says peer is not
>> suppose to register.
>>  
>> The gateway and the asterisk box are on two different location
>> (two network, 2 differrent IP address).
>>  
>> I would appreciate any kind of tutorial or advice on how to make
>> it work.
>>  
>> Thanks
>> 
>>
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>
> Hi,
> I've an installation working with Portech MV-370. I'm supposing
> it's quite similar to what you have. If it could be useful to you,
> this is my sip.conf configuration file.
>
> [GSMGtw1]
> type=friend
> context=from-gsm
> host=dynamic; we have a DHCP assigned address
> secret=reallyverysecret
> nat=no  ; there is not NAT between phone
> and Asterisk
> canreinvite=no
> dtmfmode=INFO
> insecure=invite ; required to overcome
> authentication problems in incoming calls
> call-limit=1   ; permit only 1 outgoing call
> at a time
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> qualify=500
>
> I remember that I've found a bug on the firmware that prevents to
> the unit to register correctly on my asterisk box unless I'm using
> the raw IP address instead of the name of the asterisk box. I
> remember something wrong in cryptography chiper/dechiper based on
> realm... So, if you have problems, let's try to specify the
> asterisk raw IP address in the Portech.
>
> Best regards,
> Marco Signorini.
>
>

Hi,

I don't know if the problem could be in the Mobile to Lan or Lan to
Mobile settings because these  settings are related on how calls coming
from/to mobile are routed.  I didn't use the Portech routing features at
all because I need a simple GSM gateway to/from the asterisk box.
For this reason:
1. The only rule I've on Mobile to Lan is CID=*; url=...@192.168.0.5
where "mob" is the extension I've generated in the asterisk box under
the context where the Portech operates;
2. The only rule I've on Lan to Mobile is URL=*; Call Num=#

I think the most relevant parameters for your problem are under the
"Service Domain" menu option (assuming that the firmware you have is
similar to what I've). On this menu I've compiled the 1st Realm (as I've
only one account) like that:

UserName: GSMGtw1
RegisterName: GSMGtw1
RegisterPassword: reallyverysecret
Domain Server: 192.168.0.5
Proxy Server: 192.168.0.5

Pay attention that, having specified the Domain Server with the raw IP
address, asterisk needs to be able to authenticate peers associated to
that. For this reason I've set:

domain=192.168.0.5

on sip.conf [general] section (remember to issue a sip reload from
asterisk cli).

Hope this helps!


Best regards.
Marco Signorini




Marco Signorini
INGEGNI Tech S.r.l.
http://www.ingegnitech.com
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[asterisk-users] Snom 300 vs Grandstream gxp

[Julian Lyndon-Smith]

Can anyone who has used both comment on the pros and cons ? Need to buy 
about 30 of these, for a small company with limited IT support.

Julian

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Re: [asterisk-users] evaluate SIP response codes in dialplan

[Philipp Kempgen]

Klaus Darilion schrieb:
> Are they identical to the ISDN hangup causes?
> http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php

Yes. What you pass to Hangup() are Q.931 ISDN cause codes.
See causes.h and hangup_cause2sip() in chan_sip.c for a list.


   Philipp Kempgen

-- 
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Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
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Re: [asterisk-users] Signal Asterisk hangupcauses from one Asterisk to another

[Philipp Kempgen]

Philipp Kempgen schrieb:
> Klaus Darilion schrieb:
>> One of the problems with hangupcause is, that is might get changed from 
>> one Asterisk to another - e.g. Hangup(3) generates a SIP 404 response 
>> which gets translated to hangupcause 1. So, a mechanism to signal 
>> Asterisk hangupcauses from one Asterisk to another Asterisk would be nice.
>> 
>> IIRC I once saw a prorietary Asterisk header (X-Hangupcause or similar) 
>> in a SIP response, but I could find it currently.
> 
> asterisk 1:
> 
> SIPAddHeader(X-Hangupcause: 1);
> Dial(SIP/${ext...@asterisk2);
> 
> asterisk 2:
> 
> if ("${CHANNEL(channeltype)}" = "SIP") {
>   if ("${SIP_HEADER(X-Hangupcause)}" != "") {
>   Set(HANGUPCAUSE=${SIP_HEADER(X-Hangupcause)});

Don't know if that works. Maybe more like so:
Set(PRI_CAUSE=${SIP_HEADER(X-Hangupcause)});
Hangup(${SIP_HEADER(X-Hangupcause)});

>   }
> }

Actually the other way round.


   Philipp Kempgen

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AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
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[asterisk-users] pstn hangs up: MWI no message waiting ??

[sean darcy]

pstn incoming on a TDM400P, sometimes i* won't answer, going into
a loop like this:

  -- Starting simple switch on 'DAHDI/4-1'
[Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 
18 (Ring Begin)...
[Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 
(Ring/Answered)...
[Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: 
Channel 4 no message waiting!
 -- Hungup 'DAHDI/4-1'


I don't have any Message Waiting set ( or at least I don't think so.)

Restarting * solves it for a while.

Any suggestions?

sean


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[asterisk-users] Signal Asterisk hangupcauses from one Asterisk to another (was: Re: evaluate SIP response codes in dialplan)

[Philipp Kempgen]

Klaus Darilion schrieb:
> One of the problems with hangupcause is, that is might get changed from 
> one Asterisk to another - e.g. Hangup(3) generates a SIP 404 response 
> which gets translated to hangupcause 1. So, a mechanism to signal 
> Asterisk hangupcauses from one Asterisk to another Asterisk would be nice.
> 
> IIRC I once saw a prorietary Asterisk header (X-Hangupcause or similar) 
> in a SIP response, but I could find it currently.

asterisk 1:

SIPAddHeader(X-Hangupcause: 1);
Dial(SIP/${ext...@asterisk2);

asterisk 2:

if ("${CHANNEL(channeltype)}" = "SIP") {
if ("${SIP_HEADER(X-Hangupcause)}" != "") {
Set(HANGUPCAUSE=${SIP_HEADER(X-Hangupcause)});
}
}


   Philipp Kempgen

-- 
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Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
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Re: [asterisk-users] FWD and IPCall

[Philipp Kempgen]

David @ULC schrieb:

> Jan 15 17:18:43 ERROR[2636]: chan_sip.c:11355 handle_request: Missing Cseq.
> Dropping this SIP message, it's incomplete.

This is the important part.
One of your phones or FWD or IPCall does not speak SIP properly.


   Philipp Kempgen

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Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
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[asterisk-users] want to add SipAddHeader in call out file

[Mian M Asif]

Dear Alex Balashov and All others,
can anyone give me the example how i can add local/channel with out
call file which used for Callback, Below is my dialplan for Callback.

Need to know where i can add SipAddHeader() in below dialplan. I want
to add in call leg one.

exten => _X.,1,wait(1)
exten => _X.,2,Set(outCallerID=${exten:1})
exten => _X.,3,Busy(1)
exten => _X.,4,Hangup()

exten => h,1,GotoIf($["${InvalidUser}" = "1"]?20:2)
exten => h,2,DeadAGI(STD/STD-CBLeg1-RadAuth.pl|${SIP_HEADER(Call-ID)})
exten => h,3,Set(CALLERID(number)=${CALLERID(number)})
exten => h,4,System(echo channel:
SIP/${callback...@${lcr_terminator_std} > /tmp/${CALLERID(number)})
exten => h,5,System(echo context: STD-callback-leg2 >> /tmp/${CALLERID(number)})
exten => h,6,System(echo extension: s >> /tmp/${CALLERID(number)})
exten => h,7,System(echo priority: 1 >> /tmp/${CALLERID(number)})
exten => h,8,System(echo callerid: ${outCallerID} >>
/tmp/${CALLERID(number)}) ; Your CallerID goes here
exten => h,9,System(echo maxretries: 0 >> /tmp/${CALLERID(number)})
exten => h,10,System(echo retrytime: 3 >> /tmp/${CALLERID(number)})
exten => h,11,System(echo Set: confID=${confID} >> /tmp/${CALLERID(number)})
exten => h,12,System(echo Set: calltime=${calltime} >> /tmp/${CALLERID(number)})
exten => h,13,System(echo Set: CallBackNo=${CALLERID(number)} >>
/tmp/${CALLERID(number)})
exten => h,14,System(echo Set: Leg1CallID=${Leg1CallID} >>
/tmp/${CALLERID(number)})
exten => h,15,System(echo sleep 5 > /tmp/${CALLERID(number)}.2)
exten => h,16,System(echo mv /tmp/${CALLERID(number)}
/var/spool/asterisk/outgoing >> /tmp/${CALLERID(number)}.2)
exten => h,17,System(chmod 775 /tmp/${CALLERID(number)}.2)
exten => h,18,System(/tmp/${CALLERID(number)}.2)
exten => h,19,NoOp(Hanging up ...!!)
exten => h,20,Hangup()

[STD-callback-leg2]

exten => s,1,NoOp(Entering callback-leg2)
exten => s,2,Set(CALLERID(number)=${CallBackNo})

;-- The Script Authorizes the user on Basis of Caller ID--
;-- Plays an IVR, gets destination Phno in SIP_Dest variable -
exten => s,3,Set(TIME_NOW=${EPOCH})
exten => 
s,4,DeadAGI(STD/STD-CBLeg2-RadAuthAcc.pl|${confID}|${calltime}|${TIME_NOW}|${SIP_HEADER(Call-ID)}|${Leg1CallID})
exten => s,5,hangup()

Regards,
Asif


Date: Fri, 16 Jan 2009 06:17:35 -0500
From: Alex Balashov 
Subject: Re: [asterisk-users] want to add SipAddHeader in call out
   file
To: Asterisk Users Mailing List - Non-Commercial Discussion
   
Message-ID: <49706ccf.8040...@evaristesys.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Use a Local/ channel in the Originate command, which can punt the
outbound leg through dial plan logic that can call SipAddHeader() and
tack on the header.

Mian M Asif wrote:

> How to add SipAddHeader in outgoing call file.
> I am implementing a Callback scenario, in which a user makes a call to
> Local Access Number. The  system have to callback to the user. During
> callback a call file is generated. All I want, is to add
> SipAddHeader("pchargingvector","val") in outgoing Invite.
> How can I achieve this?
>
>
> regards,
> Asif

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Re: [asterisk-users] Portech MV-378 with Asterisk

[Pascal Bruno]

Thanks for your reply!

Can you tell me what you have in your Portech configuration settings (Mobile
to Lan Settings; Sip Proxy settings etc...)  My sip.conf file is pretty
similar to yours but still cant register.



On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini wrote:

> Emmanuel Pascal Bruno wrote:
>
>   Has anyone been able to configure portech's mv-378 gateway with
> asterisk?
>
> I did the configuration as per the manual but it does not work.
>
> My server sees the portech gateway, but when the gateway is trying to
> register to my server it fails.  It says peer is not suppose to register.
>
> The gateway and the asterisk box are on two different location (two
> network, 2 differrent IP address).
>
> I would appreciate any kind of tutorial or advice on how to make it work.
>
> Thanks
>
> --
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> Hi,
> I've an installation working with Portech MV-370. I'm supposing it's quite
> similar to what you have. If it could be useful to you, this is my sip.conf
> configuration file.
>
> [GSMGtw1]
> type=friend
> context=from-gsm
> host=dynamic; we have a DHCP assigned address
> secret=reallyverysecret
> nat=no  ; there is not NAT between phone and
> Asterisk
> canreinvite=no
> dtmfmode=INFO
> insecure=invite ; required to overcome authentication
> problems in incoming calls
> call-limit=1   ; permit only 1 outgoing call at a time
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> qualify=500
>
> I remember that I've found a bug on the firmware that prevents to the unit
> to register correctly on my asterisk box unless I'm using the raw IP address
> instead of the name of the asterisk box. I remember something wrong in
> cryptography chiper/dechiper based on realm... So, if you have problems,
> let's try to specify the asterisk raw IP address in the Portech.
>
> Best regards,
> Marco Signorini.
>
>
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Re: [asterisk-users] Asterisk Upgrade

[Torintino T]

Thanks to you. 

> Date: Fri, 16 Jan 2009 13:24:16 +
> From: gordon+aster...@drogon.net
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk Upgrade
> 
> On Fri, 16 Jan 2009, Alex Balashov wrote:
> 
> > 1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk
> 
> I'd suggest not removing /etc/asterisk if that's the only source of your 
> config files... If you (re)generate them from elsewhere, it's probably OK.
> 
> and the important one, I'd have thought is
> 
>/usr/lib/asterisk/modules
> 
> Gordon
> 
> 
> 
> >
> > 2. Install 1.2.29.
> >
> >
> > Torintino T wrote:
> >
> >> How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully
> >> again in steps please.
> >>
> >> 
> >> From: torinti...@hotmail.com
> >> To: asterisk-users@lists.digium.com
> >> Date: Fri, 16 Jan 2009 03:25:33 +0200
> >> Subject: [asterisk-users] Asterisk Upgrade
> >>
> >>
> >> I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9
> >> i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7
> >> all of the IAX trunks got not working at all.
> >>
> >> I tried to downgrade by "make clean; make; make install" in Atserisk
> >> 1.2.29 directory.but "make" gives errors in the end.
> >>
> >> How can i downgrade asterisk again and undo all changes i made?. (in
> >> steps please).
> >>
> >> and can Backup and Restore return all the previous asterisk 
> >> configurations?.
> >>
> >> Thanks.
> >>
> >> 
> >> Invite your mail contacts to join your friends list with Windows Live
> >> Spaces. It's easy! Try it!
> >> 
> >> 
> >> See all the ways you can stay connected to friends and family
> >> 
> >>
> >>
> >> 
> >>
> >> ___
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> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > -- 
> > Alex Balashov
> > Evariste Systems
> > Web: http://www.evaristesys.com/
> > Tel: (+1) (678) 954-0670
> > Direct : (+1) (678) 954-0671
> > Mobile : (+1) (678) 237-1775
> >
> > ___
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> >
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> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
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Re: [asterisk-users] Asterisk Upgrade

[Gordon Henderson]

On Fri, 16 Jan 2009, Alex Balashov wrote:

> 1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk

I'd suggest not removing /etc/asterisk if that's the only source of your 
config files... If you (re)generate them from elsewhere, it's probably OK.

and the important one, I'd have thought is

   /usr/lib/asterisk/modules

Gordon



>
> 2. Install 1.2.29.
>
>
> Torintino T wrote:
>
>> How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully
>> again in steps please.
>>
>> 
>> From: torinti...@hotmail.com
>> To: asterisk-users@lists.digium.com
>> Date: Fri, 16 Jan 2009 03:25:33 +0200
>> Subject: [asterisk-users] Asterisk Upgrade
>>
>>
>> I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9
>> i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7
>> all of the IAX trunks got not working at all.
>>
>> I tried to downgrade by "make clean; make; make install" in Atserisk
>> 1.2.29 directory.but "make" gives errors in the end.
>>
>> How can i downgrade asterisk again and undo all changes i made?. (in
>> steps please).
>>
>> and can Backup and Restore return all the previous asterisk configurations?.
>>
>> Thanks.
>>
>> 
>> Invite your mail contacts to join your friends list with Windows Live
>> Spaces. It's easy! Try it!
>> 
>> 
>> See all the ways you can stay connected to friends and family
>> 
>>
>>
>> 
>>
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> -- 
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (678) 237-1775
>
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Re: [asterisk-users] Asterisk Upgrade

[Torintino T]

Thanks.

> Date: Fri, 16 Jan 2009 07:15:29 -0500
> From: abalas...@evaristesys.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk Upgrade
> 
> 1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk
> 
> 2. Install 1.2.29.
> 
> 
> Torintino T wrote:
> 
> > How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully 
> > again in steps please.
> > 
> > 
> > From: torinti...@hotmail.com
> > To: asterisk-users@lists.digium.com
> > Date: Fri, 16 Jan 2009 03:25:33 +0200
> > Subject: [asterisk-users] Asterisk Upgrade
> > 
> > 
> > I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9
> > i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7
> > all of the IAX trunks got not working at all.
> > 
> > I tried to downgrade by "make clean; make; make install" in Atserisk 
> > 1.2.29 directory.but "make" gives errors in the end.
> > 
> > How can i downgrade asterisk again and undo all changes i made?. (in 
> > steps please).
> > 
> > and can Backup and Restore return all the previous asterisk configurations?.
> > 
> > Thanks.
> > 
> > 
> > Invite your mail contacts to join your friends list with Windows Live 
> > Spaces. It's easy! Try it! 
> > 
> > 
> > See all the ways you can stay connected to friends and family 
> > 
> > 
> > 
> > 
> > 
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> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
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[asterisk-users] Can not fetch SIP_HEADER incase of Transfer

[Krunal Patel]

Hi,

I am using Asterisk 1.4.12.1 version.

Scenario for the call is as below:

UAC  Asterisk
UAC   Transfered to
   |--Invite->|
|  |
   |
|---Invite>| |
   |   |<--302
Moved-|  |
   |
|-Invite>|


I am setting a transfer context using set(__TRANSFER_CONTEXT=outgoing) .
Now in outgoing context , I am fetching a SIP header.
But asterisk gives me warnings as below & do not returns value of sip
header.

WARNING[5410]: chan_sip.c:11678 function_sipchaninfo_read: This function can
only be used on SIP channels.
WARNING[5410]: chan_sip.c:11503 func_header_read: This function can only be
used on SIP channels.

Can you please let me know why this is happening?
Why I do not get SIP header in transfer context?

Is anybody there who have faced the same issue before?

Thanks in advance!!

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[asterisk-users] dialing trunk-to-trunk

[Leonja Cerebro]

Hello All,I'm very new in asterisk.Please help - how I can write conf files
(or some example) for to delete one ext. and to add another, it means for
example:
I need to call from one asterisk to another by trunk to trunk and my dialing
(for ex.) 100#...@1.2.1.2
when the the trunk of first asterisk is 1...@1.2.1.2 and  trunk of second
1...@1.3.1.3,
symbol '#' in this case doesn't important, I can use any another or some
symbols.
Sorry, if it is stupid, but I cannot get it.

Thanks
Leos
-- 
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Re: [asterisk-users] Asterisk Upgrade

[Alex Balashov]

1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk

2. Install 1.2.29.


Torintino T wrote:

> How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully 
> again in steps please.
> 
> 
> From: torinti...@hotmail.com
> To: asterisk-users@lists.digium.com
> Date: Fri, 16 Jan 2009 03:25:33 +0200
> Subject: [asterisk-users] Asterisk Upgrade
> 
> 
> I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9
> i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7
> all of the IAX trunks got not working at all.
> 
> I tried to downgrade by "make clean; make; make install" in Atserisk 
> 1.2.29 directory.but "make" gives errors in the end.
> 
> How can i downgrade asterisk again and undo all changes i made?. (in 
> steps please).
> 
> and can Backup and Restore return all the previous asterisk configurations?.
> 
> Thanks.
> 
> 
> Invite your mail contacts to join your friends list with Windows Live 
> Spaces. It's easy! Try it! 
> 
> 
> See all the ways you can stay connected to friends and family 
> 
> 
> 
> 
> 
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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[asterisk-users] dialing trunk to trunk

[Leonja Cerebro]

Hello All,I'm very new in asterisk.Please help - how I can write conf files
(or some example) for to delete one ext. and to add another, it means for
example:
I need to call from one asterisk to another by trunk to trunk and my dialing
(for ex.) 100#...@1.2.1.2
when the the trunk of first asterisk is 1...@1.2.1.2 and  trunk of second
1...@1.3.1.3,
symbol '#' in this case doesn't important, I can use any another or some
symbols.
Sorry, if it is stupid, but I cannot get it.

Thanks
Leos

-- 
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Re: [asterisk-users] Digium TE220 supported protocol

[Benoit]

Benoit a écrit :
> Laurent a écrit :
>   
>> Those terms would be ISDN-related. VN4 is Version Number 4, and
>> ETSI is the European standards-adopting organization for telecoms.
>> So you might want to check for "E1 support" (ISDN in Europe,
>> basically) if you want to connect a PRI-capable equipment - I
>> assume that's what you are looking for since you mentioned the TE220.
>>
>> If you read French, you might want to look at this page also:
>>
>> http://blog.nicolargo.com/2008/01/installation-dune-carte-digium-avec-asterisk.html
>>
>> I found it very useful when I installed recently a TE220 card.
>>   
>> 
>
>
> Hi,
>
> yes it's for connecting an PRI line. In fact i already have one PRI
> connected to a 'real' PRI line,provided
> by France Telecom. But this one is a little bit special, it's for
> connecting to this kind of box:
> http://www.oneaccess-net.com/en/one400.htm
>
> Which is a SDSL/PRI gateway (yes, i asked for a direct SIP/H323
> connection, but "it's not in our product line" was
> the only answer ..).
>
> Until know the L1 link doesn't goes up, so i'm looking for every bit of
> info i can use. The FT line is configured like that:
>
> span=1,1,0,ccs,hdb3
> bchan=1-15
> dchan=16
>
> And work quite well, i've duplicated this for the access thru the One400
> box:
> span=2,0,0,ccs,hdb3
> bchan=32-46
> dchan=47
>
> With no luck until now. I'm reading the one400 manual, and here are the
> defaults setup for the PRI interface:
>
>   CLI(config)# interface pri 5/0
>   CLI(config-if)# physical-interface E1
>   CLI(config-if)# framing Df
>   CLI(config-if)# linecode hdb3
>
> And here are the available framing:
>
>CLI(config-e1)# framing {none | df | mf | emf}
> Specifies the framing type.
> None: no framing. Only used for CES / unstructured mode
> Df: double frame, no CRC4. For E1 only. (Default value)
> Mf: multiframe (CRC4). For E1 only
> Emf: extended multiframe (CRC4). For E1 only.
> Sf: Super-frame (for T1 only)
> Esf: extended super frame (for T1 only)
>
> I also found this:
>
> The ONE 400 supports CCS mode (transport of
> signaling messages over a D channel). CAS (Channel Associated
> Signaling) mode is not supported.
>
>
> However there is no references to this 'double frame' mode in the
> dahdi/system.conf span configuration options
> # framing::
> #   one of 'd4' or 'esf' for T1 or 'cas' or 'ccs' for E1 and BRI.
> #  'd4' could be referred to as 'sf' or 'superframe'
>
>   

Also, i've used a straight Cat5E to connect the One400 to the TE220, both
seem to share the same Pin setup:
One 400:

1: RX (+)
2: RX (-)
4: TX (+)
5: TX (-)


TE 220:

1: RX
2: RX
4: TX
5: TX


Should i do some special crossing cable instead ? connecting Tx from one
side to other side Rx ?


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Re: [asterisk-users] Asterisk Upgrade

[Torintino T]

How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully again 
in steps please.

From: torinti...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 16 Jan 2009 03:25:33 +0200
Subject: [asterisk-users] Asterisk Upgrade









I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9

i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7

all of the IAX trunks got not working at all.

I tried to downgrade by "make clean; make; make install" in Atserisk 1.2.29 
directory.but "make" gives errors in the end.

How can i downgrade asterisk again and undo all changes i made?. (in steps 
please).

and can Backup and Restore return all the previous asterisk configurations?.

Thanks.

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It's easy! Try it!
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Re: [asterisk-users] Digium TE220 supported protocol

[Benoit]

Laurent a écrit :
> Those terms would be ISDN-related. VN4 is Version Number 4, and
> ETSI is the European standards-adopting organization for telecoms.
> So you might want to check for "E1 support" (ISDN in Europe,
> basically) if you want to connect a PRI-capable equipment - I
> assume that's what you are looking for since you mentioned the TE220.
>
> If you read French, you might want to look at this page also:
>
> http://blog.nicolargo.com/2008/01/installation-dune-carte-digium-avec-asterisk.html
>
> I found it very useful when I installed recently a TE220 card.
>   


Hi,

yes it's for connecting an PRI line. In fact i already have one PRI
connected to a 'real' PRI line,provided
by France Telecom. But this one is a little bit special, it's for
connecting to this kind of box:
http://www.oneaccess-net.com/en/one400.htm

Which is a SDSL/PRI gateway (yes, i asked for a direct SIP/H323
connection, but "it's not in our product line" was
the only answer ..).

Until know the L1 link doesn't goes up, so i'm looking for every bit of
info i can use. The FT line is configured like that:

span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16

And work quite well, i've duplicated this for the access thru the One400
box:
span=2,0,0,ccs,hdb3
bchan=32-46
dchan=47

With no luck until now. I'm reading the one400 manual, and here are the
defaults setup for the PRI interface:
   
  CLI(config)# interface pri 5/0
  CLI(config-if)# physical-interface E1
  CLI(config-if)# framing Df
  CLI(config-if)# linecode hdb3

And here are the available framing:

   CLI(config-e1)# framing {none | df | mf | emf}
Specifies the framing type.
None: no framing. Only used for CES / unstructured mode
Df: double frame, no CRC4. For E1 only. (Default value)
Mf: multiframe (CRC4). For E1 only
Emf: extended multiframe (CRC4). For E1 only.
Sf: Super-frame (for T1 only)
Esf: extended super frame (for T1 only)

I also found this:

The ONE 400 supports CCS mode (transport of
signaling messages over a D channel). CAS (Channel Associated
Signaling) mode is not supported.


However there is no references to this 'double frame' mode in the
dahdi/system.conf span configuration options
# framing::
#   one of 'd4' or 'esf' for T1 or 'cas' or 'ccs' for E1 and BRI.
#  'd4' could be referred to as 'sf' or 'superframe'




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Re: [asterisk-users] Call Stealing

[David fire]

2009/1/16 Geoff Lane 

> On Friday, January 16, 2009, ddf...@gmail.com wrote:
>
> > do you program in any language? if yes just read the chapters about
> > agi in the asterisk book  you can find it in support section in
> > www.asterisk.org
>
> I'm a reasonable PHP and VBScript programmer and have dabbled since
> the 1980s in a wide range of languages from 6502 machine code upward.
> I've no Perl but I could learn or else use PHP. So, it would be an
> interesting exercise when I can find the time!
>
> Thanks again,
>
> --
> Geoff
>
>
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the book has a very good example in php.
David


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Re: [asterisk-users] want to add SipAddHeader in call out file

[Alex Balashov]

Use a Local/ channel in the Originate command, which can punt the 
outbound leg through dial plan logic that can call SipAddHeader() and 
tack on the header.

Mian M Asif wrote:

> How to add SipAddHeader in outgoing call file.
> I am implementing a Callback scenario, in which a user makes a call to
> Local Access Number. The  system have to callback to the user. During
> callback a call file is generated. All I want, is to add
> SipAddHeader("pchargingvector","val") in outgoing Invite.
> How can I achieve this?
> 
> 
> regards,
> Asif
> 
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Evariste Systems
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Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Re: [asterisk-users] Call Stealing

[Geoff Lane]

On Friday, January 16, 2009, ddf...@gmail.com wrote:

> do you program in any language? if yes just read the chapters about
> agi in the asterisk book  you can find it in support section in
> www.asterisk.org

I'm a reasonable PHP and VBScript programmer and have dabbled since
the 1980s in a wide range of languages from 6502 machine code upward.
I've no Perl but I could learn or else use PHP. So, it would be an
interesting exercise when I can find the time!

Thanks again,

-- 
Geoff


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[asterisk-users] want to add SipAddHeader in call out file

[Mian M Asif]

How to add SipAddHeader in outgoing call file.
I am implementing a Callback scenario, in which a user makes a call to
Local Access Number. The  system have to callback to the user. During
callback a call file is generated. All I want, is to add
SipAddHeader("pchargingvector","val") in outgoing Invite.
How can I achieve this?


regards,
Asif

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Re: [asterisk-users] Call Stealing

[ddfire]

do you program in any language? if yes just read the chapters about
agi in the asterisk book  you can find it in support section in
www.asterisk.org
if you can't program send me an email I think this agi will be easy.
I will program it for you (if you can't program)

2009/1/16, Geoff Lane :
> On Thursday, January 15, 2009, David fire wrote:
>
>> hey it is preatty easy now i understand the problem
>
>> is simple
>
>> hangup in new location
>
>> dial steal code for asterisk is just an extension and it should start an
>> AGI
>
>> the system search for the call in the same group
>> bridge the channel to the current channel asterisk 1.6
>
>> or
>
>> the system search for the call in the same group (AGI)
>> send the channel to a conference (AGI search for the first free
>> conference)
>> join the current channel to the conference (AGI or AGI set a variable whit
>> the conference number)
>
> That sounds like a reasonable idea. However, I've never written an AGI
> script and so I'm not sure how a script would detect which channel to
> steal. Checking through TFOT, I see there's CHANNEL STATUS - although
> I have no idea how to use it!
>
> Thanks for the pointer.
>
> --
> Geoff
>
>
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Re: [asterisk-users] mISDN BRI Asterisk 1.4

[Lee Wilson]

Hi Francesco,

You were correct.  I pulled the cable out before everyone got in this morning 
and it was a cross over.  I've now connected a proper straight-through ISDN 
cable (don't know what the Nortel was using before) and L1 is now up on 
Asterisk:

BEGIN STACK_LIST:
  * Port 1 Type NT Prot. PTP L2Link DOWN L1Link:UP Blocked:0  Debug:1

However I'm now seeing the following message as L2 is not coming up:

P[ 1] !!! Could not Get the L2 up after 3 Attemps!!!

I checked that misdn.conf was setup to use PTMP (despite was it is saying 
above):


hfcmulti
mISDN_dsp
mISDN

1
2
3
4



Also, how can I definately tell it is running in PTMP mode? 

msisdnportinfo does not distinguish:
Port  1: NT-mode BRI S/T interface port (for phones)
 -> Interface can be Poin-To-Point/Multipoint.

Port  2: NT-mode BRI S/T interface port (for phones)
 -> Interface can be Poin-To-Point/Multipoint.


And Asterisk says it is running in PTP as well when the module is loaded:
*CLI> module load chan_misdn.so
mISDN_close: fid(21) isize(131072) inbuf(0xb7b12008) irp(0xb7b12008) 
iend(0xb7b12008)
  == Parsing '/etc/asterisk/misdn.conf': Found
P[ 0] Got: 1ptp,2ptp from get_ports
  == Registered channel type 'mISDN' (Channel driver for mISDN Support 
(Bri/Pri))
  == Registered application 'misdn_set_opt'
  == Registered application 'misdn_facility'
  == Registered application 'misdn_check_l2l1'
P[ 0] -- mISDN Channel Driver Registered --
 Loaded chan_misdn.so => (Channel driver for mISDN Support (BRI/PRI))

I've got onto the Cisco router and found the BRI interface to be configured as 
below:

interface BRI0
 no ip address
 encapsulation ppp
 dialer pool-member 1
 isdn switch-type basic-net3
 no cdp enable
 ppp authentication chap callin
end

And the following events are being reported:

Jan 16 10:03:06.099: ISDN BR0: Could not bring up interface
Jan 16 10:03:06.103: BRI0: wait for isdn carrier timeout, call id=0x8004

The router also thinks that Layer 1 is still down:
Global ISDN Switchtype = basic-net3
ISDN BRI0 interface
dsl 0, interface ISDN Switchtype = basic-net3
Layer 1 Status:
DEACTIVATED
Layer 2 Status:
Layer 2 NOT Activated
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask:  0x8003

Looks like it may be worth speaking with OpenVox regarding exactly how there 
can is setup, only problem is I'm not sure what I should be asking them.

Apologies for this long email. 

Regards

Lee

--- On Wed, 14/1/09, Francesco Peeters (linux)  wrote:

> Are you using an ISDN cross cable? I don't know these
> cards, but most cards are wired as a DTE type device (TE
> port like a router or phone) and not a DCE type device (NT
> box). So you might have Tx-Tx and Rx-Tx instead of Rx-Tx and
> Tx-Rx... ;-)
> 
> (Note that ISDN cross cables are definately NOT the same as
> a CAT5E cross cable!)
> 



  


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Re: [asterisk-users] multiple registration to sip trunking provider.

[Klaus Darilion]

I do not know cordiip thus I do not know how these 3 different accounts 
are signaled to you, but some tips:

A SIP peer is always identified by host:port - thus there is at "peer" 
level no way to differ them. But in the register command you specify the 
contact to be called, e.g. 1646H25. Thus, if you use 3 different 
contacts you should be able to differ the 3 accounts in the incoming 
context using 3 different dial patterns.

regards
klaus

Andrea Borghi schrieb:
> a strange problem of multiple sip registrations and peer selection in 
> sip.conf is calling for your suggestions!!
> 
> let's examine this scenario:
> 
> some numbers and passwords hidden with HHHs to protect the guilty :)
> 
> I have 3 distinct sip subscriptions with cordiaip.net provider in US. For 
> each of these i insert in sip.conf (with peer name differences and relefant 
> number/password differences, of course]
> 
> ---
> register => 1646H25:hh...@soft1.ny.cordiaip.net/1646H25
> 
> [cordiaus1]  
> type=friend 
> secret=H
> username=1646H25
> fromuser=1646H25
> fromdomain=soft1.ny.cordiaip.net
> host=soft1.ny.cordiaip.net
> call-limit=5
> outboundproxy=soft1.ny.cordiaip.net 
> disallow=all
> allow=gsm
> allow=alaw
> allow=ulaw
> context=DID_cordia
> insecure=port
> ---
> 
> the sip registrations are OK and all seeems fine, BUT
> i have difficulties to map the incoming call because * is making mistakes in 
> matching the incoming sip INVITE to the relevant peer.
> 
> Please note that ALL the peers share the very same host and sip port.
> 
> When i make a call to one of the subscribed cordia number, in sip debug i get 
> a packes similar to this:
> 
> 
> <--- SIP read from 38.98.115.34:5060 --->
> INVITE sip:16462487...@87.241.44.202 SIP/2.0
> Via: SIP/2.0/UDP 38.98.115.34:5060;branch=z9hG4bK22981681-bdb335
> To: 
> From: ;tag=2298168-fdb335
> Call-ID: 4926-0-1232058...@38.98.115.7
> CSeq: 1 INVITE
> Contact: 
> Server: Sansay-SIP/8.0
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: 201
> 
> v=0
> o=Sansay-SPX 11 11 IN IP4 38.98.115.9
> s=Session Controller
> c=IN IP4 38.98.115.9
> t=0 0
> m=audio 15986 RTP/AVP 0 18
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=ptime:20
> 
> 
> please note the From: and To: lines, I receive a From: with the caller CID 
> (my mobile phone, in this case) and a To: with the sip number the call is 
> directed to; this seems OK to me.
> 
> I have read the chan_sip.c source file and it seems that
> when * receives this invite, it wals through the list of sip 
> users/peers/friends to search for the correct entry from which cloning the 
> sip parameters for the channel (as moh class, call limits, codecs and such) 
> using the host IP as the key (if type=peer) or the caller number (if 
> type=user) getting the values from the From: header.
> 
> This seems very strange, because the user part of the From: header is 
> potentially ANY number and the host part (and not the port, because is is 
> always 5060 and there is insecure=port in place) in this scenario is not 
> unique due to the 3 peers definitions.
> 
> Please keep in mind that if i utilize only one registration i have absolutely 
> no problems and can configure * correctly. The problem presents itself ONLY 
> with multiple peers with multiple registrations to the same host/port.
> 
> I cannot request cordia to forward me the numbers via an unique sip 
> registration (sip trunking) because it seems that they don't offer this 
> service. (but it may well be that i hadn't asked the right question)
> 
> Can anyone suggest how to implement a correct sip trunking for this scenario, 
> in which I have the incoming calls of the three registration going in a 
> specific context (not the default, see context=DID_cordia in the peer 
> definitions) and the outgoing calls going out via a specific user (so i can 
> choose at the dialplan level with which number i am presenting myself in 
> outgoing calls)
> 
> I have spent some days trying various combinations of peers and users 
> definitions, going in all cases to crash on the wall of the algorithm * uses 
> to select the correct peer for the incoming calls.
> 
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Re: [asterisk-users] evaluate SIP response codes in dialplan

[Klaus Darilion]

Are they identical to the ISDN hangup causes?
http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php

klaus

Johansson Olle E schrieb:
> 15 jan 2009 kl. 13.02 skrev John covici:
> 
>> That is very nice, but where are the HANGUPCAUSE values documented?
> That's the issue...
> 
> include/asterisk/causes.h is a good reference for now.
> 
> /O
>>
>> Thanks.
>>
>> on Thursday 01/15/2009 Johansson Olle E(o...@edvina.net) wrote
>>> 14 jan 2009 kl. 14.02 skrev Klaus Darilion:
>>>
 Hi!

 Is it somehow possible to evaluate the SIP response code inside the
 dialplan?

 I have an Asterisk server which forwards requests to various PSTN
 gateways with SIP. If the Dial() attempt is not successful I want to
 differ at least these 3 options:
 - called destination is busy (486): e.g. activate auto-redial
 - called destination does not exist, unassigned number (404)
 - gateway is broken, error, circuit busy (e.g. 503)

 486 is mapped to DIALSTATUS=BUSY
 but both 503 and 404 is mapped to DIALSTATUS=CONGESTION

 As when Asterisk forwards the response with SIP to the caller the  
 same
 response code is used, I suspect this information must be stored
 somewhere inside the channel variable. So, are there any means to
 access it?
>>> Check the HANGUPCAUSE, it's much more detailed than DIALSTATUS.
>>>
>>> We do map the SIP (and all other protocol errors in various channel
>>> drivers) codes to ISDN hangup causes, which gives you much more
>>> information about
>>> why a call failed.
>>>
>>> The conversion we're using follows the RFC, and where that doesn't
>>> cover it, Cisco's documentation.
>>>
>>> /Olle
>>>
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>> -- 
>> Your life is like a penny.  You're going to lose it.  The question is:
>> How do
>> you spend it?
>>
>> John Covici
>> cov...@ccs.covici.com
>>
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> 
> ---
> * Olle E Johansson - o...@edvina.net
> * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
> 
> 
> 
> 
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Re: [asterisk-users] evaluate SIP response codes in dialplan

[Klaus Darilion]

Johansson Olle E schrieb:
> 15 jan 2009 kl. 12.42 skrev Klaus Darilion:
> 
>>
>> Johansson Olle E schrieb:
>>> 14 jan 2009 kl. 18.57 skrev Philipp Kempgen:
>>>
 Klaus Darilion schrieb:
> Philipp Kempgen schrieb:
>> Klaus Darilion schrieb:
>>> Is it somehow possible to evaluate the SIP response code inside  
>>> the
>>> dialplan?
>> No.
>> Part of the reasoning is that Asterisk is meant to be a multi-
>> protocol PBX, not a SIP softswitch.
> This is IMO a stupid limitation. There are dozens of ISDN cause
> codes,
> dozens of SIP response codes and similar in other protocols, but
> Dial()
> only exports BUSY or CONGESTION ..
 I know. But the developers didn't want to add it.
>>> Which is incorrect. We don't want to add expose every protocol to the
>>> dialplan if not needed. As Josh and I've stated, we have the
>>> HANGUPCAUSE that gives you this level of detail, but in a
>>> multiprotocol way.
>>>
>>> The most important feature of Asterisk is that it's a multiprotocol
>>> PBX. Even if I think there's only one protocol for the future,  
>>> there's
>>> still a lot of old stuff out there and the beauty is that I can
>>> produce services in asterisk covering all of these without knowing  
>>> the
>>> details of all these protocols. It would be really bad if I had to
>>> write one app for every protocol covered by my dialplan.
>> That's OK. HANGUPCAUSE is OK. Nevertheless a configurable mapping  
>> cause
>> codes <-> SIP response codes would be nice :-)
> Absolutely - contact me off line to discuss such a project :-)
> 
> In the meantime, we could document this a bit better.

Yes - for example a note in the documentation of DIALSTATUS which refers 
to HANGUPCAUSE.

One of the problems with hangupcause is, that is might get changed from 
one Asterisk to another - e.g. Hangup(3) generates a SIP 404 response 
which gets translated to hangupcause 1. So, a mechanism to signal 
Asterisk hangupcauses from one Asterisk to another Asterisk would be nice.

IIRC I once saw a prorietary Asterisk header (X-Hangupcause or similar) 
in a SIP response, but I could find it currently.

regards
klaus



regards
klaus

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Re: [asterisk-users] Portech MV-378 with Asterisk

[Marco Signorini]

Emmanuel Pascal Bruno wrote:
> Has anyone been able to configure portech's mv-378 gateway with asterisk?
>  
> I did the configuration as per the manual but it does not work.
>  
> My server sees the portech gateway, but when the gateway is trying to
> register to my server it fails.  It says peer is not suppose to register.
>  
> The gateway and the asterisk box are on two different location (two
> network, 2 differrent IP address).
>  
> I would appreciate any kind of tutorial or advice on how to make it work.
>  
> Thanks
> 
>
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Hi,
I've an installation working with Portech MV-370. I'm supposing it's
quite similar to what you have. If it could be useful to you, this is my
sip.conf configuration file.

[GSMGtw1]
type=friend
context=from-gsm
host=dynamic; we have a DHCP assigned address
secret=reallyverysecret
nat=no  ; there is not NAT between phone and
Asterisk
canreinvite=no
dtmfmode=INFO
insecure=invite ; required to overcome authentication
problems in incoming calls
call-limit=1   ; permit only 1 outgoing call at a time
disallow=all
allow=ulaw
allow=alaw
allow=gsm
qualify=500

I remember that I've found a bug on the firmware that prevents to the
unit to register correctly on my asterisk box unless I'm using the raw
IP address instead of the name of the asterisk box. I remember something
wrong in cryptography chiper/dechiper based on realm... So, if you have
problems, let's try to specify the asterisk raw IP address in the Portech.

Best regards,
Marco Signorini.

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Re: [asterisk-users] How to transfer a call from one Asterisk Server to another

[Lenz Emilitri]

Why don't you simply Dial() the call to a separate box keeping Asterisk out
of the audio path?

l.

2009/1/16 Paul 

>  Can anyone tell me how I can completely move an established call off of
> one Asterisk server to another?
>
> In our case we have a server with our IVR.  Depending upon digits entered,
> the call can be transferred to any of our other servers depending where the
> extension or queue reside.
> We would like to completely move the call off of the first box so we don't
> tie up resources on it.
>
> In our lab we are testing with 1.4.22.1
>
> Our provider which delivers inbound calls to us uses a Sonus gateway.   So
> far, testing has shown that if we transfer the inbound call prior to any
> media playback, it works.  But, if the IVR plays media, then it is failing,
> with a 500 internal server error being returned.
>
> Thanks for any help
>
>
>
>
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-- 
Loway - home of QueueMetrics - http://queuemetrics.com
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Re: [asterisk-users] Call Stealing

[Geoff Lane]

On Thursday, January 15, 2009, David fire wrote:

> hey it is preatty easy now i understand the problem

> is simple 

> hangup in new location

> dial steal code for asterisk is just an extension and it should start an 
> AGI 

> the system search for the call in the same group
> bridge the channel to the current channel asterisk 1.6

> or

> the system search for the call in the same group (AGI)
> send the channel to a conference (AGI search for the first free conference)
> join the current channel to the conference (AGI or AGI set a variable whit 
> the conference number)

That sounds like a reasonable idea. However, I've never written an AGI
script and so I'm not sure how a script would detect which channel to
steal. Checking through TFOT, I see there's CHANNEL STATUS - although
I have no idea how to use it!

Thanks for the pointer.

-- 
Geoff


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Re: [asterisk-users] FWD and IPCall

[David @ULC]

I am using Asterisk and
IPKall.
I tried exactly below mentioned configuration
.

http://lists.digium.com/pipermail/asterisk-users/2008-January/203607.html

But I get an Error as



== Parsing '/etc/asterisk/sip_notify.conf': Found
-- Executing NoOp("SIP/66.54.140.46-b781b470", "from-ipkall") in new stack
-- Executing NoOp("SIP/66.54.140.46-b781b470", "INTL. NUMBER/"INTL. NUMBER"
<2067770020>") in new stack
-- Executing Dial("SIP/66.54.140.46-b781b470", "Local/200 at internal") in
new stack
-- Called 200 at internal
-- Executing AGI("Local/200 at inter...@default-f232,2", "agi://
127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/200 at inter...@default-f232,2", "SIP/200 at
inter...@sip64||tTor") in new stack
-- Called 200 at inter...@sip64
-- Local/200 at inter...@default-f232,1 is ringing
Jan 15 17:18:43 ERROR[2636]: chan_sip.c:11355 handle_request: Missing Cseq.
Dropping this SIP message, it's incomplete.
== Spawn extension (default, 200 at internal, 2) exited non-zero on
'Local/200 at inter...@default-f232,2'
-- Executing DeadAGI("Local/200 at inter...@default-f232,2", "agi://
127.0.0.1:4577/call_log") in new stack
== Spawn extension (from-ipkall, 901835, 3) exited non-zero on
'SIP/66.54.140.46-b781b470'
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing DeadAGI("Local/200 at inter...@default-f232,2", "agi://
127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-0-CANCEL--)")
in new stack
-- AGI Script
agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-0-CANCEL--)
completed, returning 0
== Refreshing DNS lookups.


Any Solution ?
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