Posted: Tue Aug 31, 2004 5:44 am Post subject: [Asterisk-bsd] SIP Clients - No voice.
Hello list,
I could install and configure asterisk on my FreeBSD-5.2.1-Release :)
I create 2 extensions and my clients can call each other but
there's no sound on those calls !!!!
Why????
Someone could help me ?
Posted: Tue Aug 31, 2004 10:09 am Post subject: [Asterisk-bsd] SIP Clients - No voice.
Try invoking
set verbose 9
At the asterisk console prompt and watch the console messages as the calls
progress for more info.
Rich
Quote:
-----Original Message-----
From: asterisk-bsd-bounces@lists.digium.com
[mailto:asterisk-bsd-bounces@lists.digium.com] On Behalf Of
Jefferson Carvalho
Sent: Tuesday, August 31, 2004 8:44 AM
To: asterisk-bsd@lists.digium.com
Subject: [Asterisk-bsd] SIP Clients - No voice.
Hello list,
I could install and configure asterisk on my
FreeBSD-5.2.1-Release :) I create 2 extensions and my clients
can call each other but there's no sound on those calls !!!!
Why????
Someone could help me ?
Posted: Tue Aug 31, 2004 10:49 am Post subject: [Asterisk-bsd] SIP Clients - No voice.
Thanks Dr. Rich Murphey,
Look what i got!
apollo*CLI> set verbose 9
-- Executing Dial("SIP/1260-3e37", "SIP/1261|20|rtT") in new stack
-- Called 1261
-- SIP/1261-c412 is ringing
-- SIP/1261-c412 answered SIP/1260-3e37
-- Attempting native bridge of SIP/1260-3e37 and SIP/1261-c412
-- Attempting native bridge of SIP/1260-3e37 and SIP/1261-c412
Aug 31 15:48:34 WARNING[135442432]: chan_sip.c:673 retrans_pkt: Maximum
retries exceeded on call
9F37D538-FB63-11D8-B3A5-005004803F0B@172.20.1.133 for seqno 2746
(Non-critical Response)
The other side answer the call , the connection is done .. but i don't
have voice
on the other side!!!!!!
Regards,
-Jefferson Carvalho
Dr. Rich Murphey escreveu:
Quote:
Try invoking
set verbose 9
At the asterisk console prompt and watch the console messages as the calls
progress for more info.
Rich
>-----Original Message-----
>From: asterisk-bsd-bounces@lists.digium.com
>[mailto:asterisk-bsd-bounces@lists.digium.com] On Behalf Of
>Jefferson Carvalho
>Sent: Tuesday, August 31, 2004 8:44 AM
>To: asterisk-bsd@lists.digium.com
>Subject: [Asterisk-bsd] SIP Clients - No voice.
>
>Hello list,
>
>I could install and configure asterisk on my
>FreeBSD-5.2.1-Release :) I create 2 extensions and my clients
>can call each other but there's no sound on those calls !!!!
>Why????
>Someone could help me ?
>
>Regards,
>
>-Jefferson Carvalho
>
>_______________________________________________
>Asterisk-BSD mailing list
>Asterisk-BSD@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-bsd
>
>
>
>
Posted: Tue Aug 31, 2004 11:41 am Post subject: [Asterisk-bsd] SIP Clients - No voice.
On August 31, 2004 02:48 pm, Jefferson Carvalho wrote:
ok what codecs are you using? what type of phone ? and cut and paste one of
your basic sip setup for one of the phones....
Quote:
Thanks Dr. Rich Murphey,
Look what i got!
apollo*CLI> set verbose 9
-- Executing Dial("SIP/1260-3e37", "SIP/1261|20|rtT") in new stack
-- Called 1261
-- SIP/1261-c412 is ringing
-- SIP/1261-c412 answered SIP/1260-3e37
-- Attempting native bridge of SIP/1260-3e37 and SIP/1261-c412
-- Attempting native bridge of SIP/1260-3e37 and SIP/1261-c412
Aug 31 15:48:34 WARNING[135442432]: chan_sip.c:673 retrans_pkt: Maximum
retries exceeded on call
9F37D538-FB63-11D8-B3A5-005004803F0B@172.20.1.133 for seqno 2746
(Non-critical Response)
The other side answer the call , the connection is done .. but i don't
have voice
on the other side!!!!!!
Regards,
-Jefferson Carvalho
Dr. Rich Murphey escreveu:
>Try invoking
>
>set verbose 9
>
>At the asterisk console prompt and watch the console messages as the calls
>progress for more info.
>
>Rich
>
>>-----Original Message-----
>>From: asterisk-bsd-bounces@lists.digium.com
>>[mailto:asterisk-bsd-bounces@lists.digium.com] On Behalf Of
>>Jefferson Carvalho
>>Sent: Tuesday, August 31, 2004 8:44 AM
>>To: asterisk-bsd@lists.digium.com
>>Subject: [Asterisk-bsd] SIP Clients - No voice.
>>
>>Hello list,
>>
>>I could install and configure asterisk on my
>>FreeBSD-5.2.1-Release :) I create 2 extensions and my clients
>>can call each other but there's no sound on those calls !!!!
>>Why????
>>Someone could help me ?
>>
>>Regards,
>>
>>-Jefferson Carvalho
>>
>>_______________________________________________
>>Asterisk-BSD mailing list
>>Asterisk-BSD@lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-bsd
>
>_______________________________________________
>Asterisk-BSD mailing list
>Asterisk-BSD@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-bsd
>
>__________ NOD32 1.853 (20040830) Information __________
>
>This message was checked by NOD32 antivirus system.
>http://www.nod32.com
apollo*CLI> set verbose 9
-- Executing Dial("SIP/1260-3e37", "SIP/1261|20|rtT") in new stack
-- Called 1261
-- SIP/1261-c412 is ringing
-- SIP/1261-c412 answered SIP/1260-3e37
-- Attempting native bridge of SIP/1260-3e37 and SIP/1261-c412
-- Attempting native bridge of SIP/1260-3e37 and SIP/1261-c412
Aug 31 15:48:34 WARNING[135442432]: chan_sip.c:673 retrans_pkt: Maximum
retries exceeded on call
9F37D538-FB63-11D8-B3A5-005004803F0B@172.20.1.133 for seqno 2746
(Non-critical Response)
The other side answer the call , the connection is done .. but i don't
have voice
on the other side!!!!!!
Regards,
-Jefferson Carvalho
Dr. Rich Murphey escreveu:
>Try invoking
>
>set verbose 9
>
>At the asterisk console prompt and watch the console messages as the calls
>progress for more info.
>
>Rich
>
>>-----Original Message-----
>>From: asterisk-bsd-bounces@lists.digium.com
>>[mailto:asterisk-bsd-bounces@lists.digium.com] On Behalf Of
>>Jefferson Carvalho
>>Sent: Tuesday, August 31, 2004 8:44 AM
>>To: asterisk-bsd@lists.digium.com
>>Subject: [Asterisk-bsd] SIP Clients - No voice.
>>
>>Hello list,
>>
>>I could install and configure asterisk on my
>>FreeBSD-5.2.1-Release :) I create 2 extensions and my clients
>>can call each other but there's no sound on those calls !!!!
>>Why????
>>Someone could help me ?
>>
>>Regards,
>>
>>-Jefferson Carvalho
>>
>>_______________________________________________
>>Asterisk-BSD mailing list
>>Asterisk-BSD@lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-bsd
>
>_______________________________________________
>Asterisk-BSD mailing list
>Asterisk-BSD@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-bsd
>
>__________ NOD32 1.853 (20040830) Information __________
>
>This message was checked by NOD32 antivirus system.
>http://www.nod32.com
Posted: Tue Aug 31, 2004 11:53 am Post subject: [Asterisk-bsd] SIP Clients - No voice.
Richard,
I'm using at this first momment only X-Lite as Sip client.
I'm waiting for Sipura and Grandstream devices
arrive from USA.
I'll have to put a TDM04B ( 2FXO/2FXS ) to
work .. too. ( My PABX is a siemens HiPath )
So ... i'll have so much work..!
>>>>>>A bit of my sip.conf.
[1260]
type=friend
username=1260
mailbox=2500
context =default
secret=jeff
callerid="Jefferson Carvalho" <1260>
host=dynamic
nat=yes ; X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
;disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
Regards,
-Jefferson
Richard Neese escreveu:
Quote:
On August 31, 2004 02:48 pm, Jefferson Carvalho wrote:
ok what codecs are you using? what type of phone ? and cut and paste one of
your basic sip setup for one of the phones....
Posted: Tue Aug 31, 2004 12:18 pm Post subject: [Asterisk-bsd] SIP Clients - No voice.
Jefferson Carvalho wrote:
Quote:
Richard,
I'm using at this first momment only X-Lite as Sip client.
I'm waiting for Sipura and Grandstream devices
arrive from USA.
I'll have to put a TDM04B ( 2FXO/2FXS ) to
work .. too. ( My PABX is a siemens HiPath )
So ... i'll have so much work..!
Start with connecting everything without having NAT between clients and Asterisk.
When you have that working, start reconfiguring for the NAT...
Start simple. NAT traversal is not simple.
Quote:
>>>>>>A bit of my sip.conf.
[1260]
type=friend
username=1260
mailbox=2500
context =default
secret=jeff
callerid="Jefferson Carvalho" <1260>
host=dynamic
nat=yes ; X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
Quote:
;disallow=all
You need to enable this before you have any allow= lines! Remove the semicolon.
Quote:
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
Posted: Tue Aug 31, 2004 12:25 pm Post subject: [Asterisk-bsd] SIP Clients - No voice.
Hi Josansson,
I'm using FreeBSD 5.2.1-RELEASE / Asterisk-1.0.RC2
Those clients aren't behind nat.
They're on the same net!
Regards,
-Jefferson
Olle E. Johansson escreveu:
Quote:
Jefferson Carvalho wrote:
> Richard,
>
> I'm using at this first momment only X-Lite as Sip client.
> I'm waiting for Sipura and Grandstream devices
> arrive from USA.
> I'll have to put a TDM04B ( 2FXO/2FXS ) to
> work .. too. ( My PABX is a siemens HiPath )
> So ... i'll have so much work..!
Start with connecting everything without having NAT between clients
and Asterisk.
When you have that working, start reconfiguring for the NAT...
Start simple. NAT traversal is not simple.
>
> >>>>>>A bit of my sip.conf.
>
> [1260]
> type=friend
> username=1260
> mailbox=2500
> context =default
> secret=jeff
> callerid="Jefferson Carvalho" <1260>
> host=dynamic
> nat=yes ; X-Lite is behind a NAT router
> canreinvite=no ; Typically set to NO if behind NAT
> ;disallow=all
You need to enable this before you have any allow= lines! Remove the
semicolon.
> allow=gsm ; GSM consumes far less bandwidth than
> ulaw
> allow=ulaw
> allow=alaw
>
Posted: Tue Aug 31, 2004 12:35 pm Post subject: [Asterisk-bsd] SIP Clients - No voice.
Jefferson Carvalho wrote:
Quote:
Hi Josansson,
I'm using FreeBSD 5.2.1-RELEASE / Asterisk-1.0.RC2
Those clients aren't behind nat.
They're on the same net!
Remove nat=yes and re-instate the disallow= line.
Try again.
/O
PS. And read the docs for the SIP channel on the wiki - there's a lot of docs
over there. http://www.voip-info.org :-)
Quote:
Regards,
-Jefferson
Olle E. Johansson escreveu:
> Jefferson Carvalho wrote:
>
>> Richard,
>>
>> I'm using at this first momment only X-Lite as Sip client.
>> I'm waiting for Sipura and Grandstream devices
>> arrive from USA.
>> I'll have to put a TDM04B ( 2FXO/2FXS ) to
>> work .. too. ( My PABX is a siemens HiPath )
>> So ... i'll have so much work..!
>
>
> Start with connecting everything without having NAT between clients
> and Asterisk.
> When you have that working, start reconfiguring for the NAT...
> Start simple. NAT traversal is not simple.
>
>>
>> >>>>>>A bit of my sip.conf.
>>
>> [1260]
>> type=friend
>> username=1260
>> mailbox=2500
>> context =default
>> secret=jeff
>> callerid="Jefferson Carvalho" <1260>
>> host=dynamic
>> nat=yes ; X-Lite is behind a NAT router
>> canreinvite=no ; Typically set to NO if behind NAT
>
>
>
>> ;disallow=all
>
>
> You need to enable this before you have any allow= lines! Remove the
> semicolon.
>
>> allow=gsm ; GSM consumes far less bandwidth than
>> ulaw
>> allow=ulaw
>> allow=alaw
>>
>
> What kind of BSD system are you using?
>
> /O
>
> _______________________________________________
> Asterisk-BSD mailing list
> Asterisk-BSD@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-bsd
>
> __________ NOD32 1.853 (20040830) Information __________
>
> This message was checked by NOD32 antivirus system.
> http://www.nod32.com
>
>
>
Posted: Tue Aug 31, 2004 10:26 pm Post subject: [Asterisk-bsd] SIP Clients - No voice.
Welcome!
I got back from the rest. Does anyone can email me (if one took
notifications on bugs.digium.com) latest questions, if present for me? oej?
So, my *s lives all this time without any problems. Questions 2, are new
drivers zaptel-fbsd avail?
BR,
Oryx
----- Original Message -----
From: "Jefferson Carvalho" <jefferson@credishop.com.br>
To: "Asterisk on BSD discussion" <asterisk-bsd@lists.digium.com>
Sent: Tuesday, August 31, 2004 11:25 PM
Subject: Re: [Asterisk-bsd] SIP Clients - No voice.
Quote:
Hi Josansson,
I'm using FreeBSD 5.2.1-RELEASE / Asterisk-1.0.RC2
Those clients aren't behind nat.
They're on the same net!
Regards,
-Jefferson
Olle E. Johansson escreveu:
> Jefferson Carvalho wrote:
>
>> Richard,
>>
>> I'm using at this first momment only X-Lite as Sip client.
>> I'm waiting for Sipura and Grandstream devices
>> arrive from USA.
>> I'll have to put a TDM04B ( 2FXO/2FXS ) to
>> work .. too. ( My PABX is a siemens HiPath )
>> So ... i'll have so much work..!
>
> Start with connecting everything without having NAT between clients
> and Asterisk.
> When you have that working, start reconfiguring for the NAT...
> Start simple. NAT traversal is not simple.
>
>>
>> >>>>>>A bit of my sip.conf.
>>
>> [1260]
>> type=friend
>> username=1260
>> mailbox=2500
>> context =default
>> secret=jeff
>> callerid="Jefferson Carvalho" <1260>
>> host=dynamic
>> nat=yes ; X-Lite is behind a NAT router
>> canreinvite=no ; Typically set to NO if behind NAT
>
>
>> ;disallow=all
>
> You need to enable this before you have any allow= lines! Remove the
> semicolon.
>
>> allow=gsm ; GSM consumes far less bandwidth than
>> ulaw
>> allow=ulaw
>> allow=alaw
>>
>
> What kind of BSD system are you using?
>
> /O
>
> _______________________________________________
> Asterisk-BSD mailing list
> Asterisk-BSD@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-bsd
>
> __________ NOD32 1.853 (20040830) Information __________
>
> This message was checked by NOD32 antivirus system.
> http://www.nod32.com
>
>
>
Posted: Wed Sep 01, 2004 8:26 am Post subject: [Asterisk-bsd] SIP Clients - No voice.
Jefferson Carvalho <jefferson@credishop.com.br> writes:
Quote:
-- SIP/1261-c412 answered SIP/1260-3e37
-- Attempting native bridge of SIP/1260-3e37 and SIP/1261-c412
This means it's asking the SIP phones at either end to transmit voice
data directly between each other, instead of the data stream passing
through Asterisk.
Quote:
The other side answer the call , the connection is done .. but i
don't have voice on the other side!!!!!!
...and this would be the situation, if the direct link between the two
phones failed to work, for instance because of a firewall between them
that didn't permit the direct connection -- or because of NAT trouble.
-tih
--
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway
www.eunet.no T: +47-22092958 M: +47-93013940 F: +47-22092901
Posted: Wed Sep 08, 2004 8:40 am Post subject: [Asterisk-bsd] SIP Clients - No voice.
On 01/09/2004 03:53 Jefferson Carvalho said the following:
Quote:
I'll have to put a TDM04B ( 2FXO/2FXS ) to
work .. too. ( My PABX is a siemens HiPath )
are the freebsd zaptel drivers for the E100P or the TE405P working on
either 4.10 or 5.x ?
--
Regards, /\_/\ "All dogs go to heaven."
dinesh@alphaque.com (0 0) http://www.alphaque.com/
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