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How To Auto Answer A Sip Phone

Hi,


i wanted to make dial plan in such a way that the any incoming call to the sip phone should auto answer.(auto pickup) . Help.




regards Upendra

Issue With A Ticket System Subscribed To Asterisk-users

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Thanks,

Chan_sip Sending From Wrong Source, Address When Multiple Interfaces Are Used

This can and will happen, if the multi-interface Asterisk server is also being used as the router between these networks.

In this case, the packet sent by the client system to 192.168.1.1, is being sent by the client to the client’s active (possibly default/only)
routing gateway, which will be 10.0.2.1. The packet arrives at the server/router, and because it matches the IP address assigned to one of the server’s network adapters (and Asterisk is bound to all of them)
it’s delivered to Asterisk.

When Asterisk replies, it’s doing so via a socket which is bound to 0.0.0.0. Since there’s no specific IP address bound to this socket, the kernel has to pick one of the host’s IP addresses to put into the packet it sends… and the one it picks is the one assigned to the network adapter on which it chooses to transmit the port. In this case, that will be 10.0.2.1.

So, the client sends a packet to 192.168.1.1 and gets a response from 10.0.2.1, which is a huge WTF situation for many protocols.

This is not just an issue for SIP. I’ve had exactly this same problem with IAX2 clients and Asterisk, and had to apply the exact same cure – I tell IAX2 to bind itself only to my host’s externally-routable public IP address, and not to
0.0.0.0. Then, if I specify the public IP address as the server in each IAX2 client configuration, everything works fine.

This is probably a not-unusual configuration if you’re setting up a modest-size “VoIP-only” or “VoIP-mostly” network on a budget… a Linux or other Unix-ish system running Asterisk will often have enough CPU power to handle the RTP routing between networks, saving you the cost of a dedicated router. I have this very issue on my own system – a modest home network with a couple of internal LANs, some IP ranges set aside for VPNs of various sorts, and one externally routable IP address.


Would be well worthwhile… and if you can port similar code over into chan_iax2, it would fix the problem there as well.

Chan_sip Sending From Wrong Source Address When Multiple Interfaces Are Used

“Kevin P. Fleming” writes:


Imagine you have a server with two interfaces, eth0 with 192.168.1.1/24
and eth1 with 10.0.2.1/24. Further imagine that you wish to be able to move phones between the networks without changing the SIP server address, so you set 192.168.1.1 as the SIP server no matter which network they happen to be on.

Now the phones which happen to be connected to eth1 will send a request to 192.168.1.1. If Asterisk is bound to 0.0.0.0, the reply will come from 10.0.2.1. This could be solved if Asterisk did a connect() to the socket and use the same socket for answering. That would tell the system IP stack that this is in fact a connection, and so the system would ensure that the reply source IP would be correct.

Alas, few programmers are aware that you can even do connect() for UDP, and I believe it would be a rather large change to the Asterisk SIP
stack to pass connected sockets around rather than just remembering IP
addresses and port numbers. (Admittedly I haven’t looked at that code in ages, so I could easily be wrong).

The workaround is to explicitly bind to 192.168.1.1. Since Asterisk can bind to precisely one address, that kills off IPv6.


/Benny

Asterisk With OpenBTS And Mobile Phone

Hello people,

I want to connect Asterisk with OpenBTS and make a call with a mobile phone.

I use:
Ubuntu 11.10 + Kernel 3.0.22
GnuRadio 3.3.0
Asterisk 1.8.13
OpenBTS 2.8
Nokia Mobile Phone

OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also OpenBSC is working with Asterisk successfully (OpenBSC is another project). Perhaps you can help me because I think it is an issue with Asterisk.

I have tried both contexts, [open-bts] and [sip_external] and both don’t work. If I want to call the mobile phone (6201) with a Twinkle soft phone (6000) I get following message in the CLI-window from Asterisk:

*CLI> sip show peers
*CLI> sip show peer 123456789101112

Asterisk log file (path: /var/log/asterisk/cdr-csv/Master.csv):

If you need more informations write me and I will give you. It would be very appreciated if some of you can help me or has an idea how I can fix this erorr. Best regards and thanks for helping.

Weird Dect Beheaviour Multiple Handsets

I have this very specific problem with two dect sets. Problem that I
have is one-way audio, in this very rare situation.

I am calling with a Siemens N510 with C610 handset to Panasonic KX-TGP500 with KX-TPA50 handset. This gives me problems when I am calling to a SIP account that is configured to ring all handsets. Then when one handset answers, I only hear the panasonic, but they don’t hear me.

When I call to an extension that is configured to ring only one handset, I don’t have this problem.

When I use panasonic on both sides, or Tiptel -> Panasonic, I don’t have this problem as well.

I am breaking my head what this could be. Any idea’s?