QUESTION sipbroker.com Asterisk12

Will Longo

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Hello everyone, i'll start off by saying, i'm not using pbxinaflash or freepbx or anything. I do frequently build systems based on the platform. I'm reaching out here because of the friendly community and expertise!

At home i have a vanilla asterisk 12 setup running on centos7, everything is working perfectly with pjsip except sipbroker. my google voice is great, trunk to my bud's cisco cm, everything! sipbroker in the logs LOOKS like its working, the calls just never go through.

relevant pjsip.conf
Code:
[sipbroker]
type=endpoint
transport=transport-udp
context=unauthenticated
disallow=all
allow=ilbc
allow=g722
allow=ulaw
allow=gsm
aors=sipbroker
 
[sipbroker]
type=aor
contact=sip:204.11.194.10:5060
 
[sipbroker]
type=identify
endpoint=sipbroker
match=204.11.194.10

relevant dialplan
Code:
exten => _*.,1,Dial(PJSIP/sipbroker/sip:${EXTEN}@sipbroker.com)
exten => 18002255288,1,Dial(PJSIP/*${EXTEN}@sipbroker)

both of these dialplans will look like they're going through but hang indefinitely. I dont see errors in any logs, but am willing to post any logs that may be useful.. has anyone got sipbroker to work in asterisk12?

it's a pretty big deal for me. I have a few users who are in afghanistan and in order to keep the regular trunk open for calls to their families, i try to send all the business type toll-free calls through sipbroker. since this isn't working, their calls are now tying up the trunks. any help would be greatly appreciated! thanks in advance and keep up the great work everyone!
 

Will Longo

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I did some additional troubleshooting with no luck.

Code:
[sipbroker]
type=registration
transport=transport-udp
outbound_auth=sipbroker
server_uri=sip:sipbroker.com
client_uri=sip:[email protected]
contact_user=driz
retry_interval=60
forbidden_retry_interval=600
expiration=3600
 
[sipbroker]
type=auth
auth_type=userpass
password=i<3wardmundy
username=piaffan

now i know my password is here in the open but please dont abuse my trust!

anyway, even with this setup, authentication fails. it would appear that treating a trunk as a peer is no longer an option with pjsip which may be the ultimate issue here. has anyone in the piaf community come up with anything to work around something like this?
 

Trimline2

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May 23, 2013
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First, edit your password right now! We are a cozy family, but hackers will see this. Yes they do read these messages.... :mad:

I have Sipbroker as a Custom Outbound Trunk, meaning it is not setup like a regular SIP trunk. The custom dial string reads:

SIP/[email protected]
 

Will Longo

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Trimline,
I was just kidding, my password isn't really i love ward mundy... or is it?

thanks for providing a tip, im actually using pjsip so, currently it doesnt appear that you can just send a number out to a destination like a URI.. it appears that you're required to have it listed as an endpoint of some sort in pjsip.conf or else you get this

Code:
[Sep 24 21:29:25] ERROR[18310]: chan_pjsip.c:1726 request: Unable to create PJSIP channel - endpoint 'sipbroker' was not found

using something like this
Code:
Dial("PJSIP/dznet-p-00000005", "PJSIP/sipbroker/sip:*[email protected]:5060")
with a pjsip.conf of this
Code:
[sipbroker]
type=endpoint
transport=transport-udp
context=unauthenticated
disallow=all
allow=ilbc
allow=g722
allow=ulaw
allow=gsm
aors=sipbroker
force_rport=yes
direct_media=no
 
[sipbroker]
type=aor
contact=sip:204.11.194.10:5060
will result in no ring and no connection, with this in the log
Code:
    -- Executing [999@driz:1] Dial("PJSIP/dznet-p-00000005", "PJSIP/sipbroker/sip:*[email protected]:5060") in new stack
    -- Called PJSIP/sipbroker/sip:*[email protected]:5060
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/dznet-p-00000005' status is 'CHANUNAVAIL'

i verified through my asterisk 11 system that the line does indeed work, it's just the pjsip config IMO.



basically from what i can tell, with pjsip, dialing direct by IP or hostname no longer functions without a corresponding endpoint entry in pjsip.conf. so if i want to call my mom arbitrarily without an ep specifically for her destination by dialing [email protected] i will get an error about a lack of endpoint even if sip would have successfully placed the call.
 

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