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greyvoip at yahoo.com.au
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 Posted: Sat Feb 02, 2008 3:59 pm Post subject: [asterisk-users] Asterisk 1.6 - Problems with SIP/REFER |
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| Quote: | ----- Original Message ----
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| Quote: | From: Jake Wicke <jake at nxtphase.net>
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| Quote: | To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com>
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| Quote: | Sent: Friday, 1 February, 2008 5:34:12 PM
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| Quote: | Subject: [asterisk-users] Asterisk 1.6 - Problems with SIP/REFER
I am having issues with transfers (SIP/REFER) using Asterisk 1.6. You will find the SIP debug below.
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| Quote: | There are three phones in this setup. 5253 and 5258 are Aastra 53i telephones, 101 is a standard phone connected through an Audiocodes gateway. All phones are registered in context ?phones? and are set to not allow reinvites. All phones can dial each other directly. The dialplan looks as follows:
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Exten => 5253,1,Dial(SIP/5253,10)
Exten => 5878,1,Dial(SIP/5878,10)
Exten => 101,1,Dial(SIP/101 at audiocodes,10)
| Quote: | Transfer fails regardless of the order (101 calls 5878, 5878 transfers to 5253 or 5878 calls 5253, 5253 transfers to 101, etc)
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| Quote: | I do not understand the message ?Spawn Extension (phones, 101, 0) exited non-zero? in the debug ? there is no ?priority zero? in a dialplan ? priority should start at 1. What is this message telling me?
What do I need to do to allow these phones to transfer calls between each other? Any help is greatly appreciated!
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Hi Jake,
I don't have the answer but I did look at your trace and something about the way the transfer is being done from your phones is not quite right. You're calling 5253, then calling 5878 and then requesting a blind transfer of 5253 to 5878. However at that stage 5878 is already on the phone. I suspect the transfer should be being requested as an attended one not a blind one.
Regards,
Greyman.
Get the name you always wanted with the new y7mail email address.
www.yahoo7.com.au/y7mail |
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oej at edvina.net
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| Posted: Sat Feb 02, 2008 4:27 pm Post subject: [asterisk-users] Asterisk 1.6 - Problems with SIP/REFER |
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1 feb 2008 kl. 18.34 skrev Jake Wicke:
| Quote: | I am having issues with transfers (SIP/REFER) using Asterisk 1.6.
You will find the SIP debug below.
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When you have issues, it's always a good idea to check the bug
tracker. There might be other people having the same issues, in some
cases, there's also a solution. If you don't find it on first search,
make sure you also search in resolved and closed issues.
For this case, we do have open bug reports. The same issue exists in
1.4 and we're working on a solution. Stay tuned.
Have a nice weekend!
/Olle |
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