[Asterisk-Users] Telasip config problem/question

Jim Lynch jimlynch1 at gmail.com
Tue May 2 09:26:56 MST 2006


I seem to be getting a connection from telasip but instead of dialing my
default extension, nothing happens.  I listen to dead air.

I have a fxo card configured and working on both inbound and outbound
calls.  Telasip is working outbound.  I put in the recommended (by telasip)
changes to the trunk for incoming, e.g.
host=gw4.telasip.com
insecure=very
qualify=yes
type=user
context=from-pstn

Then I went to incoming routes and set it up for any did, any cid, but the
telasip connection is taking a different route.  Here are the log entries
for both.

Telasip:

May  2 11:11:55 DEBUG[2670] chan_sip.c: Checking SIP call limits for device
jlynch
May  2 11:11:55 DEBUG[2670] chan_sip.c: build_route: Contact hop: <
sip:7707190068 at 4.79.19.59>
May  2 11:11:55 VERBOSE[6343] logger.c:     -- Executing
Set("SIP/jlynch-cf63", "FROM_DID=6782280738") in new stack
May  2 11:11:55 VERBOSE[6343] logger.c:     -- Executing
Answer("SIP/jlynch-cf63", "") in new stack
May  2 11:11:55 VERBOSE[6343] logger.c:     -- Executing
PlayTones("SIP/jlynch-cf63", "ring") in new stack
May  2 11:11:55 DEBUG[6343] channel.c: Scheduling timer at 160 sample
intervals
May  2 11:11:55 VERBOSE[6343] logger.c:     -- Executing
NVFaxDetect("SIP/jlynch-cf63", "0") in new stack
May  2 11:11:55 DEBUG[6343] app_nv_faxdetect.c: Preparing detect of fax
(waitdur=4ms, sildur=1000ms, mindur=100ms, maxdur=-1ms)
May  2 11:11:55 DEBUG[2670] chan_sip.c: Stopping retransmission on '
21710733271a1e37775bc8ad1259e42f at 4.79.19.59' of Response 102: Match Found
May  2 11:11:55 DEBUG[2670] chan_sip.c: Stopping retransmission on '
21710733271a1e37775bc8ad1259e42f at 4.79.19.59' of Response 103: Match Found
May  2 11:11:55 DEBUG[6343] channel.c: Scheduling timer at 0 sample
intervals
May  2 11:11:55 DEBUG[6343] app_nv_faxdetect.c: Got hangup

Zap:
May  2 10:55:09 DEBUG[2670] chan_sip.c: build_route: Contact hop: <
sip:200 at 192.168.2.99>
May  2 10:55:09 VERBOSE[6287] logger.c:     -- SIP/200-9b14 answered Zap/1-1
May  2 10:55:09 DEBUG[6287] chan_zap.c: Requested indication -1 on channel
Zap/1-1
May  2 10:55:09 DEBUG[6287] channel.c: Scheduling timer at 0 sample
intervals
May  2 10:55:15 DEBUG[6287] channel.c: Didn't get a frame from channel:
SIP/200-9b14
May  2 10:55:15 DEBUG[6287] channel.c: Bridge stops bridging channels
Zap/1-1 and SIP/200-9b14
May  2 10:55:15 DEBUG[6287] chan_sip.c: update_call_counter(200) - decrement
call limit counter
May  2 10:55:15 DEBUG[6287] app_dial.c: Exiting with DIALSTATUS=ANSWER.
May  2 10:55:15 VERBOSE[6287] logger.c:   == Spawn extension (macro-dial, s,
10) exited non-zero on 'Zap/1-1' in macro 'dial'
May  2 10:55:15 VERBOSE[6287] logger.c:   == Spawn extension (macro-dial, s,
10) exited non-zero on 'Zap/1-1' in macro 'exten-vm'
May  2 10:55:15 VERBOSE[6287] logger.c:   == Spawn extension (macro-dial, s,
10) exited non-zero on 'Zap/1-1'

I even attempted to add an incoming  route using the Telasip did, but it
didn't work either.  I have the radio button clicked in each of them to
route the call to extension 200.
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