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[asterisk-users] Dropped call on new CISCO router for no reason!


 
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symack at gmail.com
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PostPosted: Mon Jan 06, 2014 9:28 am    Post subject: [asterisk-users] Dropped call on new CISCO router for no rea Reply with quote

Hello Everyone,

Just getting in a new cisco router, and would really like to get it up and running as soon
as possible. Everything is configured from what we can see. This is a NAT setup.
After 2 seconds on a successfully established call we reach retrans max, and asterisk
disconnects the call. We have no idea why this is happening. SIP and RTP is flowing as
expected.


Your help is greatly appreciated,


Nick.
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EWieling at nyigc.com
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PostPosted: Mon Jan 06, 2014 9:38 am    Post subject: [asterisk-users] Dropped call on new CISCO router for no rea Reply with quote

This is a classic symptom of having reinvites and/or direct media enabled on Asterisk or SIP ALG enabled on the router.

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nick Cameo
Sent: Monday, January 06, 2014 9:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dropped call on new CISCO router for no reason!

Hello Everyone,

Just getting in a new cisco router, and would really like to get it up and running as soon as possible. Everything is configured from what we can see. This is a NAT setup.
After 2 seconds on a successfully established call we reach retrans max, and asterisk disconnects the call. We have no idea why this is happening. SIP and RTP is flowing as expected.

Your help is greatly appreciated,

Nick.

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symack at gmail.com
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PostPosted: Mon Jan 06, 2014 10:39 am    Post subject: [asterisk-users] Dropped call on new CISCO router for no rea Reply with quote

Hello Eric, I knew this problem all so well however, never knew CISCO sip alg was enabled bydefault. The following settings got us up and going shortly after the email:


no ip nat service sip udp port 5060


ip nat inside source static udp 192.168.2.5 5060 interface Dialer0 5060



access-list 130 permit udp any any range 8000 65535
route-map voip-rtp permit 1
match ip address 130

ip nat inside source static <PRIVATE IP> <PUBLIC IP> route-map voip-rtp



Happy New Year to All,


Nick from Toronto.
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paul.belanger at polyb...
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PostPosted: Mon Jan 06, 2014 1:33 pm    Post subject: [asterisk-users] Dropped call on new CISCO router for no rea Reply with quote

On 14-01-06 09:27 AM, Nick Cameo wrote:
Quote:
Hello Everyone,

Just getting in a new cisco router, and would really like to get it up and
running as soon
as possible. Everything is configured from what we can see. This is a NAT
setup.
After 2 seconds on a successfully established call we reach retrans max,
and asterisk
disconnects the call. We have no idea why this is happening. SIP and RTP is
flowing as
expected.

Your help is greatly appreciated,

Nick.



Show us the problem, give us a SIP trace[1].

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

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_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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