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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: fpausp on February 18, 2008, 10:34:03 PM
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Hi all,
I like to ask some questions/problems i have, i use sme 7.3 with sail v584 and a few linksys spa942 phones.
My 1st problem is i have two Trunkslines, one Trunkline is a AnalogFXO and the other one is a Sipprovider in Austria (ahool...),
when i call out via the AnalogFXO i have to use Numbers without coutry code (02626...), when i like to use the Siptrunk i must use it (00432626...), how can i combine that ?
My 2nd question is i like to see the inbound numbers on my spa942, at the moment i have always asterisk unknown, how can i fix this ?
Is it possible to use a central phonebook for multiple spa942 phones ?
regards
fpausp
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Hello fpausp
...when i like to use the Siptrunk i must use it (00432626...), how can i combine that ?
In the transformation mask for your SIP trunk put 00: 0:43 :00. This should allow you to dial regular national numbers (like 02626....) on your SIP trunk. You should also be able to dial non Austrian international numbers (such as 0044 and 0049).
at the moment i have always asterisk unknown, how can i fix this ?
You either do not have caller-id enabled for your analogue line;
or
your analogue FXO card does not recognise Austrian PTT CLID
or
you do not have zapata.conf set correctly to recognise Austrian CLID.
I'm sure that there will be an Austrian Asterisk user community who can advise you on this.
Is it possible to use a central phonebook for multiple spa942 phones ?
...Not within SAIL. The spa-942 can hold up to 100 numbers in its directory but I don't believe there is an easy way to hold a cebtral directory (as there is with Aastra and SNOM devices).
Kind Regards
S
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Hi,
Thanks for your answers, transformation mask is working. After a call with my Phonecompany the inbound callerid is also working
(callerid service is disabled per default and costs 1 Euro/month).
Thanks again
regards
fpausp
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Hi,
Some more questions please, i can see some asterisk ports with:
I like to do sailtosail with a friend, wie both use sail v584. I have setup a trunkline on each side (Privileged:yes) and a route _X6000 for me
and _X5000 for him, the connection icons are ok but when i call 6000 i got a peep peep ....
Maybe i must open some ports, netstat shows me:
[root@server ~]# netstat -npl | grep asterisk
tcp 0 0 127.0.0.1:5038 0.0.0.0:* LISTEN 11971/asterisk
tcp 0 0 0.0.0.0:2000 0.0.0.0:* LISTEN 11971/asterisk
udp 0 0 0.0.0.0:2727 0.0.0.0:* 11971/asterisk
udp 0 0 0.0.0.0:4520 0.0.0.0:* 11971/asterisk
udp 0 0 0.0.0.0:5060 0.0.0.0:* 11971/asterisk
udp 0 0 0.0.0.0:4569 0.0.0.0:* 11971/asterisk
unix 2 [ ACC ] STREAM HÖRT 2543833 11971/asterisk /var/run/asterisk/asterisk.ctl
what ports must be accessible for sailtosail, i think only iax ?
And my last question is about fax/fax2mail, is it possible without extra hardware only via my sip-provider ?
regards
fpausp
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To call your friend on 6000 you need a route dial plan of _6XXX (not _X6000).
For your friend to call 5000 he needs a route dialplan of _5XXX
IN both cases, UDP 4569 needs to be open and forwarded to the SAIL server. SAIL opens 4569 on the sme server firewall automatically but if you are running behind another firewall then you need to ensure the ports are open. I think this must be OK because your IAX link is showing "connected" icon.
You cannot FAX easily over SIP. It will usually work over a LAN but it won't work over a WAN unless you use a special FAX protocol called T38. Asterisk currently has little or no support for T38.
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I am sorry, of course it is _6XXX, that was a mistype. The routerfirewall is disabled on both sides. When i call 6000 it looks like this on my friends side:
server*CLI>
[Feb 29 23:39:27] NOTICE[5223]: chan_iax2.c:7344 socket_process: Rejected connect attempt from 212.183.xxx.xxx, who was trying to reach '6000@'
server*CLI>
and on my side:
server*CLI>
[Feb 29 21:43:43] WARNING[5241]: chan_iax2.c:7554 socket_process: Call rejected by 213.33.xxx.xxx: No authority found
server*CLI>
must i change something with username/password ?
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what does "iax2 show peers" give on each machine?
Best
S
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on my side:
server*CLI> iax2 show peers
Name/Username Host Mask Port Status
server6000/se 213.33.xxx.xxx (S) 255.255.255.255 4569 (T) OK (19 ms)
5010/fax (Unspecified) (D) 255.255.255.255 0 UNKNOWN
5009/reini (Unspecified) (D) 255.255.255.255 0 UNKNOWN
5008/horst (Unspecified) (D) 255.255.255.255 0 UNKNOWN
5007/rudolf (Unspecified) (D) 255.255.255.255 0 UNKNOWN
5006/franz (Unspecified) (D) 255.255.255.255 0 UNKNOWN
5005/ap003 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
5004/ap002 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
5003/ap001 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
9 iax2 peers [1 online, 8 offline, 0 unmonitored]
on his side:
server*CLI> iax2 show peers
Name/Username Host Mask Port Status
server5000/se 212.183.xxx.xxx (S) 255.255.255.255 4569 OK (59 ms)
6005/ap003 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
6004/ap002 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
6003/ap001 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
4 iax2 peers [1 online, 3 offline, 0 unmonitored]
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Are 'server5000' and 'server6000' the real hostnames of the machines? (config show SystemName).
Best
S
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hallo,
the systemname on both sides is: SystemName=server
iax2 show peers from my friends side:
server*CLI> iax2 show peers
Name/Username Host Mask Port Status
serverfpausp/se 212.183.xxx.xxx (S) 255.255.255.255 4569 OK (43 ms)
6005/ap003 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
6004/ap002 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
6003/ap001 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
4 iax2 peers [1 online, 3 offline, 0 unmonitored]
iax2 show peers from my side:
server*CLI> iax2 show peers
Name/Username Host Mask Port Status
serverdjfjfd/se 212.183.xxx.xxx (S) 255.255.255.255 4569 (T) OK (24 ms)
5010/fax (Unspecified) (D) 255.255.255.255 0 UNKNOWN
5009/reini (Unspecified) (D) 255.255.255.255 0 UNKNOWN
5008/horst (Unspecified) (D) 255.255.255.255 0 UNKNOWN
5007/rudolfopa (Unspecified) (D) 255.255.255.255 0 UNKNOWN
5006/franzopa (Unspecified) (D) 255.255.255.255 0 UNKNOWN
5005/ap003 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
5004/ap002 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
5003/ap001 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
9 iax2 peers [1 online, 8 offline, 0 unmonitored]
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OK, ..... The name you choose when you create the iax sibling (sail-to-sail) must be the true SystemName of the remote system. The two systems cannot have the same SystemName so you will need to change one of them if they are both the same. Change the SystemName by logging into the linux console and running the command "console". Delete your sail-to-sail trunks and redefine them. Your link will then work.
Kind Regards
S
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Hi,
Yes and No - It is working basically but i have additional problems:
1.) We can dial us vize versa but voice is going only in one direction (from me to him, i can not hear his voice but music on hold is working).
2.) We have two dynamic domains, after a while i think when the ip canges we can not reach/dial us vize versa.
regards
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We can't work magic.
If the ip address/domain name doesn't point to the server or the firewall is not forwarding and flowing 4569UDP then it won't work. Nothing to do with asterisk or SAIL, it just won't work.
If you can hear MOH but not his voice then there is aproblem flowing SIP across his machine. Make sure that localnet is set correctly in sip.conf headers and that he doesn't have any internal firewall issues.
Kind Regards
S
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Hi selintra,
I take my hat off to you ! - It is working :wink:
The last bit is to update the trunkline when my or the other server is changeing the ip to be accessible.
regards
fpausp
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Hi All,
I like to call my server/phones from outside my lan without register a client. I want to use a linux-client with ekiga softphone. Inside my lan i can call any extension without registration, i just type 5000@servername but from outside this is not working.
Any hints ?
regards
fpausp
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I like to call my server/phones from outside my lan without register a client
I guess the first question is why do you want to do this?
If you really do need to do it then you are going to have to manage NAT traversal yourself and this is not trivial when using SIP and RTP to move traffic back and forth.
S
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Hi,
Excuse my English maybe it was a bad explanation ?
Other People should be able to call me via Internet without any registration (not on my server or other carriers), they have only a pc with internet-connection, i dont like to register ther clients on my server. They should just open a Softwareclient like EKiga and make a connection to me.
Kind
fpausp
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It's not easy to do what you want to do unless the phones have their own public IP addresses or you use some mechanism for NAT traversal. Asterisk, and most carriers, use a technique called Symmetrical RTP with a NAT heartbeat/keep-alive. You can also use STUN servers or a remote SIP server with Session Border Control (i.e. a carrier). Some softphones have mechanisms that can help but the techniques need to be the same on both sides.
Good luck
Best
S
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We just use ext@domainname it seems to be working fine here..
Works better than incoming calls from some sip providers here in oz actually.
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Yes, on a quick test the phones are ringing but i had no audio, that is maybe a prob on the pc out of my lan.
regards
fpausp