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Don Pobanz

2004-Jun-14 10:41 UTC

[Asterisk-Users] Sipura 2000 not answering em_w calls

I recently purchased a Sipura 2000 and connected a phone to it which is 
connected to my asterisk box via sip.

Calls to the Sipura 2000 work fine from another sip device connected 
through *, from either an fxo or fxs (via adtran channel bank connected 
to a T400P card) port. However, when a call comes in from the phone 
company over a T1 with em_w trunks, the phone on the Sipura will ring 
but I can not answer it. With Sip debug set I will see a bunch of 
messages but the following messages seems important:

7 headers, 18 lines
Jun 14 11:35:08 WARNING[4101]: chan_sip.c:2012 find_call: Call missing 
call ID from '192.168.50.119'

(192.168.50.119 is the IP of my Sipura device)


Calls coming in from the phone company though em_w trunks work fine 
when terminated to analog phones (fxo_ks via a channel bank) or to 
another pbx analog trunks (fxs_ks via a channel bank) or to a 
Grandstream sip phone.

So, I do not know whether this is a Sipura 2000 problem or an * 
problem.

Does anyone have any light to shine on the subject.

Asterisk CVS-HEAD-06/14/04-09:03:15 built by root@localhost.localdomain 
on a i686 running Linux
Sipura 2000 software version 1.0.33

Don Pobanz

mattf

2004-Jun-14 11:37 UTC

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[Asterisk-Users] Sipura 2000 not answering em_w calls

Well, I don't think it's the sipura. We have 45 SPA-2000 adapters
connected
to 82 analog phones going through 3 asterisk servers all using E&M Wink
start T1s, and we have no problems with inbound or outbound on any of the
sipura adapter-connected phones.

Post your sip.conf entry for your sipura device, as well as what line
features you have activated on your sipura configuration.

MATT---

-----Original Message-----
From: Don Pobanz [mailto:dpobanz@hastingsutilities.com]
Sent: Monday, June 14, 2004 1:41 PM
To: 'asterisk-users@lists.digium.com'
Subject: [Asterisk-Users] Sipura 2000 not answering em_w calls


I recently purchased a Sipura 2000 and connected a phone to it which is 
connected to my asterisk box via sip.

Calls to the Sipura 2000 work fine from another sip device connected 
through *, from either an fxo or fxs (via adtran channel bank connected 
to a T400P card) port. However, when a call comes in from the phone 
company over a T1 with em_w trunks, the phone on the Sipura will ring 
but I can not answer it. With Sip debug set I will see a bunch of 
messages but the following messages seems important:

7 headers, 18 lines
Jun 14 11:35:08 WARNING[4101]: chan_sip.c:2012 find_call: Call missing 
call ID from '192.168.50.119'

(192.168.50.119 is the IP of my Sipura device)


Calls coming in from the phone company though em_w trunks work fine 
when terminated to analog phones (fxo_ks via a channel bank) or to 
another pbx analog trunks (fxs_ks via a channel bank) or to a 
Grandstream sip phone.

So, I do not know whether this is a Sipura 2000 problem or an * 
problem.

Does anyone have any light to shine on the subject.

Asterisk CVS-HEAD-06/14/04-09:03:15 built by root@localhost.localdomain 
on a i686 running Linux
Sipura 2000 software version 1.0.33

Don Pobanz
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Don Pobanz

2004-Jun-18 15:03 UTC

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[Asterisk-Users] Sipura 2000 not answering em_w calls

Issue has been resolved.

The short answer:
The format of caller ID information on em_w trunks (maybe zap channels 
too) in zapata.conf can make it so the sipura will not anwser a call.
> Calls to the Sipura 2000 work fine from another sip device connected
> through *, from either an fxo or fxs (via adtran channel bank
> connected to a T400P card) port. However, when a call comes in from 
the phone> company over a T1 with em_w trunks, the phone on the Sipura will ring
> but I can not answer it.
> Calls coming in from the phone company though em_w trunks work fine
> when terminated to analog phones (fxo_ks via a channel bank) or to
> another pbx analog trunks (fxs_ks via a channel bank) or to a
> Grandstream sip phone.
>
>
>On Monday, June 14, 2004 1:38 PM, mattf [SMTP:mattf@vicimarketing.com] 
wrote:>> Well, I don't think it's the sipura. We have 45 SPA-2000
adapters
>> connected
>> to 82 analog phones going through 3 asterisk servers all using E&M
>> Wink
>> start T1s, and we have no problems with inbound or outbound on any 
of>> the
>> sipura adapter-connected phones.
>
I finally figured out the problem. The problem was that my callerID 
information in zapata.conf
was in the format of
  callerid = "DID trunk1 <4024623600>"   instead of the format
  callerid = "DID trunk1"<4024623600>
Notice the location of the quotation marks (").

CallerID has worked fine on all other phones so it never dawned on me 
that this could be a callerID problem.

Thanks mattf for pointing out that your sipuras were working which made 
me figure it had to be one of my configurations.

Don Pobanz

Frank

2004-Jul-12 12:56 UTC

head link

[Asterisk-Users] VoiceMail - Catching caller hangup

I am trying to do some cleanup after a voice mail is left.  But if the
caller just hangs up (user does not press # to send/complete the message
or wait for timeout to be reached), the voicemail seems to do an
immediate hangup and does not step through the rest of the context.

In the case below, priority 3 never gets executed if the caller just
hangs up after speaking their voice message.

exten => s,1,ResponseTimeout(30)
exten => s,2,VoiceMail(${ARG1}${ARG2})
exten => s,3,GoToIf($[${ARG3} = 0]?s|5)

how can I trap this condition so that I can still get priority 3
executed after voicemail exits?

usedcanon

2004-Jul-12 15:57 UTC

head link

[Asterisk-Users] VoiceMail - Catching caller hangup

When the caller hangsup, asterisk will jump to extension h if exists. 

Umar.

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Frank
Sent: 12 July 2004 20:56
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] VoiceMail - Catching caller hangup

I am trying to do some cleanup after a voice mail is left.  But if the
caller just hangs up (user does not press # to send/complete the message
or wait for timeout to be reached), the voicemail seems to do an
immediate hangup and does not step through the rest of the context.

In the case below, priority 3 never gets executed if the caller just
hangs up after speaking their voice message.

exten => s,1,ResponseTimeout(30)
exten => s,2,VoiceMail(${ARG1}${ARG2})
exten => s,3,GoToIf($[${ARG3} = 0]?s|5)

how can I trap this condition so that I can still get priority 3
executed after voicemail exits?

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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