[asterisk-users] Cisco 7940 no outgoing audio

[Simon Alman]

Hi Salvatore

The firmware is PS03-08-2-00.

Unfortunately I can only packet capture on the asterisk server itself,
but I am seeing:

P 172.16.8.22 > 172.16.8.1: ICMP 172.16.8.22 udp port 2224 unreachable,
length 36
IP 172.16.8.20 > 172.16.8.1: ICMP 172.16.8.20 udp port 17099
unreachable, length 36
IP 172.16.8.20 > 172.16.8.1: ICMP 172.16.8.20 udp port 17099
unreachable, length 36
IP 172.16.8.1 > 172.16.8.20: ICMP 172.16.8.1 udp port 17228 unreachable,
length 208

Where x.1 is asterisk and x.20 is the cisco and x.22 is a polycom test
phone. We also see these errors on our working network (asterisk 1.0.10)
so they are possibly a red herring.

I suspect RTP issues but am unsure how to proceed as the Cisco phones do
not seem to allow rtp debugging via their console.

For reference our rtp.conf is:

[general]
rtpstart=10000
rtpend=20000
rtpchecksums=yes (have tried with no and made no difference)

Regards

Simon

Salvatore Giudice wrote:
> You should get a packet capture of both cisco-cisco and
> grandstream/polycom-cisco. Compare the SDP's. The cisco phone may not be
> able to understand the other vendor's devices. BTW, what version of firmware
> are you running on the cisco phones?
>
> --------------------------------------------------
> Salvatore Giudice
> Salvatore.Giudice at VoIPSecurityTraining.com
>
> VoIP Security Training, LLC
> http://VoIPSecurityTraining.com
>
> 848 N. Rainbow Blvd. #1676
> Las Vegas, NV 89107
> Phone: (617) 959-7625
> Fax: (214) 279-2906
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Simon Alman
> Sent: Tuesday, May 01, 2007 11:27 AM
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] Cisco 7940 no outgoing audio 
>
> Hi All
>
> We have a private network setup (no nat) with three types of phones
> connecting to asterisk via SIP. We have Polycom, Grandstream and Cisco
> 7940 IP phones.
>
> When we ring polycom to grandstream or grandstream to polycom then both
> phones can send and receive voice fine and all is well.
>
> When we dial any combination of Cisco and either Polycom, or Granstream
> the Cisco, no voice is being sent but the Cisco can receive voice from
> the remote phone fine.
>
> When we dial Cisco to Cisco it all works fine.
>
> I am at a loss to figure this out and any help pointing me in the right
> direction would be appreciated. We are running an old Asterisk server
> with version 1.0.10 (yeah we know) and the same mix of hardware and
> configs works fine.
>
> On the new (problem) setup we are running Asterisk 1.4.2 and our Cisco
> firmware is 08-2-00.
>