brekeke PBX can not call outside through mediatrix 1204
Moderator: Brekeke Support Team
brekeke PBX can not call outside through mediatrix 1204
1. Brekeke Product Name and version: Brekeke PBX, Version 2.1.6.6 , Pro Evaluation
2. Java version: jre1.5.0_06
3. OS type and the version: windows xp
4. UA (phone), gateway or other hardware/software involved:x-lite, mediatrix 1204
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
6. Your problem:
I bought mediatrix 1204 fxo gateway. i followed brekeke configuration on this website http://www.brekeke-sip.com/wiki/index.p ... i=0b0151a2
documentation is not saying any thing else.
i have two extension 1001 and 1002 i created a user 1204 and forward it to auto attendant. when i call from outside the gateway sends the call to 1204 and auto attendant ask for other extension and user can dial 1001 or 1002. its working fine.
for outgoing when i made ARS
matching pattern to: sip:9(.+)@
deploy pattern:sip:$1@172.16.12.111
pbx ip is 172.16.12.120
when i call 9406222 from extension 1001 it gives me error "temporarily unavailable" but i can see the in use LED is on.
the log is saying
04:15:22 AM 04:15:22 AM 1001 missed "94062222"<sip:94062222@17... "1001"<sip:1001@172.16.12.... 00:00:00
04:15:25 AM 04:17:55 AM 1204 callee "3330001" <sip:3330001@172... <sip:ivr1204@127.0.0.1:250... 04:15:25 AM 00:02:29
caller sip:1204@172.16.12.120:506... sip:3330001@172.16.12.120:... 04:15:25 AM 00:02:29
even i run ethereal there is sip 480 error temporarily unavailable.
can some one help me in this matter.
i am planning to buy PBX for 40 users only if every thing works fine.
i really appreciate
2. Java version: jre1.5.0_06
3. OS type and the version: windows xp
4. UA (phone), gateway or other hardware/software involved:x-lite, mediatrix 1204
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
6. Your problem:
I bought mediatrix 1204 fxo gateway. i followed brekeke configuration on this website http://www.brekeke-sip.com/wiki/index.p ... i=0b0151a2
documentation is not saying any thing else.
i have two extension 1001 and 1002 i created a user 1204 and forward it to auto attendant. when i call from outside the gateway sends the call to 1204 and auto attendant ask for other extension and user can dial 1001 or 1002. its working fine.
for outgoing when i made ARS
matching pattern to: sip:9(.+)@
deploy pattern:sip:$1@172.16.12.111
pbx ip is 172.16.12.120
when i call 9406222 from extension 1001 it gives me error "temporarily unavailable" but i can see the in use LED is on.
the log is saying
04:15:22 AM 04:15:22 AM 1001 missed "94062222"<sip:94062222@17... "1001"<sip:1001@172.16.12.... 00:00:00
04:15:25 AM 04:17:55 AM 1204 callee "3330001" <sip:3330001@172... <sip:ivr1204@127.0.0.1:250... 04:15:25 AM 00:02:29
caller sip:1204@172.16.12.120:506... sip:3330001@172.16.12.120:... 04:15:25 AM 00:02:29
even i run ethereal there is sip 480 error temporarily unavailable.
can some one help me in this matter.
i am planning to buy PBX for 40 users only if every thing works fine.
i really appreciate
1 2008-04-05 23:02:48.905508 172.16.12.120 172.16.12.111 SIP/SDP Request: INVITE sip:4062222@172.16.12.111, with session description
2 2008-04-05 23:02:49.005492 172.16.12.111 172.16.12.120 SIP Status: 100 Trying
4 2008-04-05 23:02:52.301157 172.16.12.111 172.16.12.120 SIP/SDP Request: INVITE sip:1204@172.16.12.120:5060, with session description
5 2008-04-05 23:02:52.301874 172.16.12.120 172.16.12.111 SIP Status: 100 Trying
6 2008-04-05 23:02:52.312486 172.16.12.111 172.16.12.120 SIP Status: 480 Temporarily Unavailable
7 2008-04-05 23:02:52.315274 172.16.12.120 172.16.12.111 SIP Request: ACK sip:4062222@172.16.12.111
8 2008-04-05 23:02:52.321428 172.16.12.120 172.16.12.111 SIP/SDP Status: 200 OK, with session description
9 2008-04-05 23:02:52.460751 172.16.12.111 172.16.12.120 SIP Request: ACK sip:1204@172.16.12.120:5060
10 2008-04-05 23:02:52.472823 172.16.12.120 172.16.12.111 RTP PT=ITU-T G.711 PCMU, SSRC=0x69BAD, Seq=453, Time=867840
pbx ip: 172.16.12.120
mediatrix: 172.16.12.111
following is the call log on pbx:
11:02:48 PM 11:02:48 PM 1001 missed "94062222"<sip:94062222@17... "1001"<sip:1001@172.16.12.... 00:00:00
11:02:52 PM 11:05:21 PM 1204 callee "3330001" <sip:3330001@172... <sip:ivr1204@127.0.0.1:250... 11:02:52 PM 00:02:28
caller sip:1204@172.16.12.120:506... sip:3330001@172.16.12.120:... 11:02:52 PM 00:02:28
2 2008-04-05 23:02:49.005492 172.16.12.111 172.16.12.120 SIP Status: 100 Trying
4 2008-04-05 23:02:52.301157 172.16.12.111 172.16.12.120 SIP/SDP Request: INVITE sip:1204@172.16.12.120:5060, with session description
5 2008-04-05 23:02:52.301874 172.16.12.120 172.16.12.111 SIP Status: 100 Trying
6 2008-04-05 23:02:52.312486 172.16.12.111 172.16.12.120 SIP Status: 480 Temporarily Unavailable
7 2008-04-05 23:02:52.315274 172.16.12.120 172.16.12.111 SIP Request: ACK sip:4062222@172.16.12.111
8 2008-04-05 23:02:52.321428 172.16.12.120 172.16.12.111 SIP/SDP Status: 200 OK, with session description
9 2008-04-05 23:02:52.460751 172.16.12.111 172.16.12.120 SIP Request: ACK sip:1204@172.16.12.120:5060
10 2008-04-05 23:02:52.472823 172.16.12.120 172.16.12.111 RTP PT=ITU-T G.711 PCMU, SSRC=0x69BAD, Seq=453, Time=867840
pbx ip: 172.16.12.120
mediatrix: 172.16.12.111
following is the call log on pbx:
11:02:48 PM 11:02:48 PM 1001 missed "94062222"<sip:94062222@17... "1001"<sip:1001@172.16.12.... 00:00:00
11:02:52 PM 11:05:21 PM 1204 callee "3330001" <sip:3330001@172... <sip:ivr1204@127.0.0.1:250... 11:02:52 PM 00:02:28
caller sip:1204@172.16.12.120:506... sip:3330001@172.16.12.120:... 11:02:52 PM 00:02:28
Hi,
I didn't use this gateway before, the following suggestion maybe not right.
From your packets, it seems Mediatrix sent the call back to pbx .
Could you set "call direction" as only for incoming calls at the following page,
http://www.brekeke-sip.com/wiki/index.p ... i=14278458
or you can open another port for outgoing calls
I didn't use this gateway before, the following suggestion maybe not right.
From your packets, it seems Mediatrix sent the call back to pbx .
Could you set "call direction" as only for incoming calls at the following page,
http://www.brekeke-sip.com/wiki/index.p ... i=14278458
or you can open another port for outgoing calls
-
- Posts: 9
- Joined: Sat Jan 05, 2008 6:14 am
Re: brekeke PBX can not call outside through mediatrix 1204
I also have the same to same problem. If you find the solution out side the Fourm please inform me, I will be thankful.
Also it is a request to expert for help....... appriciation
Also it is a request to expert for help....... appriciation
-
- Posts: 9
- Joined: Sat Jan 05, 2008 6:14 am
-
- Posts: 528
- Joined: Tue Sep 20, 2005 9:10 am
- Location: Tannersville, Pennsylvania
Awais,
The variable SipTransportQValue needs to have a number in it to register. Use a number less than 1 like .23 . It's really hard to find info about the Mediatrix out there but this variable needs to have a value for it to register. Once it registers the Brekeke configuration is simple.
Here is a link to a configuration for a similar unit that register to a sip server. Maybe you can see something you missed.
http://www.abptech.com/support/qa/index ... x-register
The variable SipTransportQValue needs to have a number in it to register. Use a number less than 1 like .23 . It's really hard to find info about the Mediatrix out there but this variable needs to have a value for it to register. Once it registers the Brekeke configuration is simple.
Here is a link to a configuration for a similar unit that register to a sip server. Maybe you can see something you missed.
http://www.abptech.com/support/qa/index ... x-register
-
- Posts: 1
- Joined: Tue Apr 06, 2010 7:59 am
- Location: United Kingdom
Translator for foru
Hello Guys. Is there a good intrenet translator for translating this forum in Swedish? I would show It to some people which don't speak english.