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[asterisk-users] Doing asteriksk with a sip trunk

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ngom at numericap.com
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Posted: Thu Mar 31, 2016 10:26 am Post subject: [asterisk-users] Doing asteriksk with a sip trunk

Hello !
I ask if it is necessary to install DAHDI and LIBPRI if we want to connect our asterisk to an operator SIP (trunk SIP).
Someone for helping me.
thanks !!!

Quote:

Le 31 mars 2016 à 15:59, Roel van Meer <roel@1afa.com> a écrit :

Ethy H. Brito writes:

Quote:

Ifconfig output looks like this:root@communiceer:~# ifconfig eth1
eth1 Link encap:Ethernet HWaddr b4:99:ba:a9:3e:e5
inet addr:x.x.x.x Bcast:x.x.x.127 Mask:255.255.255.128
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:5967421 errors:0 dropped:21425 overruns:0 frame:0

Did you notice this ............................^^^^^ value???

Yes, I did. But I assumed (hmm) that this was caused because this server is
not IPv6-enabled. See also here: http://stackoverflow.com/a/30703716/3172389

Quote:

Should not be a problem since you are complaining abou TX packets, not RX,
but...

does dmesg say anything about this?

Nope, nothing at all.

But what I can do is set the interface into promiscuous mode with tcpdump,
then there should be no dropped packets at all, I think. I'll check to make
sure.

Thanks for the heads up, and thanks for thinking with me everyone!

Cheers,

Roel

Quote:

TX packets:6085933 errors:0 dropped:0 overruns:0 carrier:0collisions:0 txqueuelen:1000
RX bytes:1223605260 (1.1 GiB) TX bytes:2096293903 (1.9 GiB)
Interrupt:17 Memory:fbfe0000-fc000000

I was thinking maybe there's a problem with the transmit queue, but 1000 is
the default value for txqueuelen and I have never needed to change it.

I have the default queueing discipline:

root@communiceer:~# tc qdisc show dev eth1
qdisc pfifo_fast 0: root refcnt 2 bands 3 priomap 1 2 2 2 1 2 0 0 1 1 1 1

Quote:

1 1 1

The output of ethtool also looks good:

root@communiceer:~# ethtool eth1
Settings for eth1:
Supported ports: [ TP ]
Supported link modes: 10baseT/Half 10baseT/Full
100baseT/Half 100baseT/Full
1000baseT/Full
Supports auto-negotiation: Yes
Advertised link modes: 10baseT/Half 10baseT/Full
100baseT/Half 100baseT/Full
1000baseT/Full
Advertised pause frame use: No
Advertised auto-negotiation: Yes
Speed: 1000Mb/s
Duplex: Full
Port: Twisted Pair
PHYAD: 1
Transceiver: internal
Auto-negotiation: on
MDI-X: on
Supports Wake-on: pumbg
Wake-on: g
Current message level: 0x00000007 (7)
drv probe link
Link detected: yes

And the nic stats also look good:

root@communiceer:~# ethtool -S eth1
NIC statistics:
rx_packets: 6071960
tx_packets: 6189424
rx_bytes: 1244435132
tx_bytes: 2117335817
rx_broadcast: 293751
tx_broadcast: 193
rx_multicast: 29827
tx_multicast: 0
rx_errors: 0
tx_errors: 0
tx_dropped: 0
multicast: 29827
collisions: 0
rx_length_errors: 0
rx_over_errors: 0
rx_crc_errors: 0
rx_frame_errors: 0
rx_no_buffer_count: 0
rx_missed_errors: 0
tx_aborted_errors: 0
tx_carrier_errors: 0
tx_fifo_errors: 0
tx_heartbeat_errors: 0
tx_window_errors: 0
tx_abort_late_coll: 0
tx_deferred_ok: 0
tx_single_coll_ok: 0
tx_multi_coll_ok: 0
tx_timeout_count: 0
tx_restart_queue: 0
rx_long_length_errors: 0
rx_short_length_errors: 0
rx_align_errors: 0
tx_tcp_seg_good: 37559
tx_tcp_seg_failed: 0
rx_flow_control_xon: 0
rx_flow_control_xoff: 0
tx_flow_control_xon: 0
tx_flow_control_xoff: 0
rx_csum_offload_good: 3447739
rx_csum_offload_errors: 2
rx_header_split: 0
alloc_rx_buff_failed: 0
tx_smbus: 0
rx_smbus: 0
dropped_smbus: 0
rx_dma_failed: 0
tx_dma_failed: 0
rx_hwtstamp_cleared: 0
uncorr_ecc_errors: 0
corr_ecc_errors: 0
tx_hwtstamp_timeouts: 0

So I really don't know where to look elsewhere..

Thanks,

Roel

Quote:

-----Original Message-----
From: Roel van Meer <roel@1afa.com>
Date: Thu, 31 Mar 2016 14:10:48
To: <dovid@telecurve.com>; Asterisk Users Mailing List - Non-Commercial
Discussion<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Lost outgoing SIP packets

Dovid Bender writes:

Quote:

The tcpdump that you are running is on the Asterisk box or via port
mirroring?

It's on the asterisk box itself.

I've already replaced the network card - no change.

Thanks,

Roel

Quote:

Regards,

Dovid

-----Original Message-----
From: Roel van Meer <roel@1afa.com>
Sender: asterisk-users-bounces@lists.digium.comDate: Thu, 31 Mar 2016
13:34:51
To: <asterisk-users@lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] Lost outgoing SIP packets

Hi list!

I have a problem where SIP packets sent by Asterisk do not hit the

wire,

Quote:

and

I don't know what could cause this.

I'm running Asterisk 1.8.28_cert5 with full SIP debug. At the same

time,

Quote:

I'm

doing a tcpdump of the traffic on the network interface. I can see in
the

SIP

Quote:

debug log that asterisk is sending packets. Most of the time, I can see
those packets in the tcpdump, as you would expect.
However, sometimes Asterisk sends a packet that *does not show up* in

the

Quote:

tcpdump. Asterisk then does several retransmits (that also don't show

up).

Quote:

The next packet that is not a retransmit does show up again.

This causes Asterisk to log the peer it was sending packets to

temporarily

Quote:

as Lagged or unreachable.

There is no outgoing firewall on this box.

Could anyone give me some pointers where to look?

If Asterisk logs "VERBOSE[13019] chan_sip.c: Reliably Transmitting

(NAT)

Quote:

to x.x.x.x:" you would expect to see that packet in a tcpdump trace,

right? What could cause this not to be so? Are there network

statistics I

Quote:

could look at? Is there a counter in /proc or /sys for problems with

sending packets? Anything?

If more information is necessary please do let me know.

Thanks a lot in advance,

Roel

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Mamadou NGOM
Ingénieur Télécommunications & Réseaux
Mobile: 06 72 45 23 03
Skype: Mamadou Numericap
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asterisk_list at earth...
Guest

Posted: Thu Mar 31, 2016 10:59 am Post subject: [asterisk-users] Doing asteriksk with a sip trunk

On Thursday 31 Mar 2016, Mamadou NGOM wrote:

Quote:

Hello !
I ask if it is necessary to install DAHDI and LIBPRI if we want to connect
our asterisk to an operator SIP (trunk SIP). Someone for helping me.
thanks !!!

No.

DAHDI is a library for hardware interfaces to POTS, ISDN and mobile lines.
LibPRI is a library specifically for primary rate ISDN interfaces (one Primary
Rate ISDN = 30 lines).

If you are connecting only SIP phones to your Asterisk server, and it is
talking to the outside world only via a SIP trunk, then you do not need DAHDI
or LibPRI.

--
AJS

Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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