[asterisk-users] Reliable SIP Trunk Provider

[Jake Wicke]

I'm wondering if any other Asterisk users have a recommendation for a reliable 
SIP Trunk provider that supports Asterisk and offers decent support.

I've worked with Coredial, Broadvox, and Broadvoice and have had some bad 
experiences with each of these providers.

Broadvoice offers low cost service, however I have constant issues with 
Broadvoice blocking my customers due to Asterisk "registering too often".  
Support either does not respond to e-mails, hangs up on phone calls, or gives 
me the "we don't support Asterisk and we can use your account no problem using 
the SIP phone on our desk" line.

Coredial resigned me into a two year agreement after making a change to my SIP 
trunk configuration without my knowledge, then demanded two years of the full 
monthly charge when I tried to cancel over a dispute regarding services that I 
did not order.  Check out coredialhorrorstory.com for the whole story.  While 
the service is decent, the customer service leaves much to be desired.

Broadvox has been the best provider that I have found so far, however I 
initially had a lot of issues with sales quoting a product which could not be 
provisioned and also not being able to deliver service on a timely schedule.  I 
also was given the run around by customer service recently on a simple request 
to add a DID number to an account.

Thanks for your input!


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[asterisk-users] SendDTMF not Working - Possible Echo Cancelling Issues

[Jake Wicke]

I am having issues with SendDTMF on an Asterisk box using Asterisk 1.4.18 and 
Zaptel 1.4.8.  As far as I can see, the issues seem to be related to echo 
cancellation.  The box has a TDM2400p installed and is using the HPEC echo 
canceller.

The problems occur when I attempt to do an outcall.  The outcall dials a pager 
number, then connects the dialed call to a context in my extensions.conf which 
sends a number of DTMF digits to the pager number.

I have changed the pager number to my telephone number and can hear a large 
amount of variance in the DTMF digits. I believe this variance is what is 
causing the problem when the pager is dialed.

I have attempted to change the toneduration in zaptel.conf - this setting seems 
to not work at all - any changes made to this setting make no difference in the 
duration of the tone sent over the channel.  The output of the script seems to 
be better (the DTMF is more clear) when echocancelwhenbridged is set to yes.

How do I change the DTMF duration on a zap channel using SendDTMF() as 
toneduration does nothing?  Is there any way to disable echo cancellation for a 
single call?

Jake Wicke





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[asterisk-users] Asterisk 1.6 - Problems with SIP/REFER

[Jake Wicke]

I am having issues with transfers (SIP/REFER) using Asterisk 1.6.  You will 
find the SIP debug below.

There are three phones in this setup.  5253 and 5258 are Aastra 53i telephones, 
101 is a standard phone connected through an Audiocodes gateway.  All phones 
are registered in context "phones" and are set to not allow reinvites.  All 
phones can dial each other directly.  The dialplan looks as follows:

[phones]
Exten => 5253,1,Dial(SIP/5253,10)
Exten => 5878,1,Dial(SIP/5878,10)
Exten => 101,1,Dial(SIP/[EMAIL PROTECTED],10)

Transfer fails regardless of the order (101 calls 5878, 5878 transfers to 5253 
or 5878 calls 5253, 5253 transfers to 101, etc)

I do not understand the message "Spawn Extension (phones, 101, 0) exited 
non-zero" in the debug - there is no "priority zero" in a dialplan - priority 
should start at 1.  What is this message telling me?

What do I need to do to allow these phones to transfer calls between each 
other?  Any help is greatly appreciated!

Here is the debug:

*CLI>   == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/5253-0823eab0", "SIP/5878") in 
new stack
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
Audio is at 10.7.10.1 port 19968
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.7.10.51:5060:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK38d90448;rport
Max-Forwards: 70
From: "5253" ;tag=as05a48c1a
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0-beta2
Remote-Party-ID: "5253" ;privacy=off;screen=no
Date: Wed, 30 Jan 2008 01:12:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 864806723 864806723 IN IP4 10.7.10.1
s=Asterisk PBX 1.6.0-beta2
c=IN IP4 10.7.10.1
t=0 0
m=audio 19968 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called 5878

<--- SIP read from UDP://10.7.10.51:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 
10.7.10.1:5060;branch=z9hG4bK38d90448;rport=5060;received=10.7.10.1
From: "5253" ;tag=as05a48c1a
To: ;tag=694417843
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, 
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Call-Info: ;appearance-index=1
Contact: 5878 
Server: Aastra 53i/2.1.0.2145
Content-Length: 0


<->
--- (12 headers 0 lines) ---
-- SIP/5878-08250098 is ringing

<--- SIP read from UDP://10.7.10.51:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
10.7.10.1:5060;branch=z9hG4bK38d90448;rport=5060;received=10.7.10.1
From: "5253" ;tag=as05a48c1a
To: ;tag=694417843
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, 
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Call-Info: ;appearance-index=1
Contact: 5878 
Server: Aastra 53i/2.1.0.2145
Supported: timer, replaces
Content-Type: application/sdp
Content-Length: 313

v=0
o=MxSIP 0 0 IN IP4 10.7.10.51
s=SIP Call
c=IN IP4 10.7.10.51
t=0 0
m=audio 3000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:ZXJ1UmhNLDFmQGNHYGAnRlpKbjEudk9Gfjh8blo/
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<->
--- (14 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.7.10.51:3000
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.7.10.51:3000
list_route: hop: 
set_destination: Parsing  for address/port to send 
to
set_destination: set destination to 10.7.10.51, port 5060
Transmitting (NAT) to 10.7.10.51:5060:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK6476d991;rport
Max-Forwards: 70
From: "5253" ;tag=as05a48c1a
To: ;tag=694417843
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0-beta2
Remote-Party-ID: "5253" ;privacy=off;screen=no
Content-Length: 0


---
-- SIP/5878-08250098 answered SIP/5253-0823eab0
-- Packet2Packet bridging SIP/5253-0823eab0 and SIP/5878-08250098

<--- SIP read from UDP://10.7.10.51:5060 --->
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.10.51:5060;branch=z9hG4bKd79bb4c662d65595a
Max-Forwards: 70
From: ;tag=694417843
To: "5253" ;tag=as05a48c1a
Call-ID: [EMAIL PROTECTED]
CSeq: 20367 INVITE
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, 
SUBSCRIBE, INFO
Allow-Eve