Re: Asterisk configuration
Can anyone recommend an asterisk hardware vendor?
I a looking for an analog 2FXS 2FXO pci card to play with asterisk and
freeswitch.
Any opinions about digium/sangoma?
I currently have an ATA grandstream 488 which I am not so happy with.
Meir Michanie
www.riunx.com
- Original message -
From: Eran Levy [EMAIL PROTECTED]
To: IGLU List linux-il@cs.huji.ac.il
Sent: Sun Oct 21 2007 12:36:42 PM IST
Subject: Re: Asterisk configuration
Tzafrir hi,
Thanks, it helped to clarify some of my mistakes.
I'm still configuring the Asterisk box - not easy at all...
Anyway, after finishing the configuration process, I will post an arranged
message to the list. I'm sure it gonna help to many users.
Eran
On 10/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Oct 17, 2007 at 11:27:59AM +0200, Ohad Levy wrote:
the PRI context
[from-pri]
X_.,1,answer
X_.,n,dial(SIP/[EMAIL PROTECTED])
This must be a copy/paste error. I figure you meant:
[from-pri]
_X.,1,answer
_X.,n,dial(SIP/[EMAIL PROTECTED])
(_X. rather than X_.). A _ at the beginning of the extension name
means it should be parsed as a pattern extension. This gives special
meaning to X, . and others.
So the above mean:
1. First-off answer the call
2. Now attempt to connect it to a specific SIP destination.
Automatically answering an incoming call may not be the best idea.
Suppose that the destination phone was busy. You still told the PRI
provider that the call was accepted, and hence the call will be charged.
Dial will answer calls when it is able to connect it with the
destination, so an explicit Answer() is not required.
Ido already gave some comments about a better usage of SIP peers rather
than mearly [EMAIL PROTECTED], so I'll assume you have a peer set up
for that destination, called dest-peer. In that case you could use:
Dial(SIP/dest-peer)
instead. This allows a better definition of codecs and such. You may
also want to dial an explicit number to the destination. For example:
Dial(SIP/dest-peer/${EXTEN})
(this dials there the number that was dialed to you, in case you have
more than one)
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il || a Mutt's
[EMAIL PROTECTED] || best
ICQ# 16849754 || friend
=
To unsubscribe, send mail to [EMAIL PROTECTED] with
the word unsubscribe in the message body, e.g., run the command
echo unsubscribe | mail [EMAIL PROTECTED]
--
Thanks,
Eran
Re: Asterisk configuration
Meir Michanie wrote: Can anyone recommend an asterisk hardware vendor? I a looking for an analog 2FXS 2FXO pci card to play with asterisk and freeswitch. Any opinions about digium/sangoma? For what it's worth, I've used both Digium and Sangoma cards in the past in productions systems and while both have their quirks, I can say I was pleased with both of them. The Sangoma cards are a little bit more hassle to set up and less commonplace, but their tech support was so superb that it did not bother at all. And I mean the CEO sending email to let me know that they fixed the rare firmware bug I ran into on Sunday morning and having their tech support following on the morning after even though I bought just a single card kind of superb support. If PCI is not a must, you might also consider getting the USB2 connected Xorcom extension banks. Haven't tried them myself but I've heard good things about them and as they are Israeli developed (some of the developers are on this list as a matter of fact) I would certainly recommend giving them a shot. One tip though: if it's a real production system and not something to play at home with, make sure to get hardware echo cancellation. Costs a little bit more, but really helps with the quality. I currently have an ATA grandstream 488 which I am not so happy with. The big plus of Grandstream equipment is that they are very cheap. This i, however, their only plus (but I'm using a Grandstream desk phone myself) :-) Hope this helps, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] IL: +972.3.7515563 ext. 201 | Fax:+972.3.7515503 US: +1.212.2026643 ext. 201 | Cel: +972.52.8260388 There once was a virtualization coder, Whose patches kept getting older, Each time upstream would drop, His documentation would slightly rot, SO APPLY MY F*$KING PATCHES OR I'LL KEEP WRITING LIMERICKS. -- Rusty Russel = To unsubscribe, send mail to [EMAIL PROTECTED] with the word unsubscribe in the message body, e.g., run the command echo unsubscribe | mail [EMAIL PROTECTED]
Re: Asterisk configuration
Tzafrir hi,
Thanks, it helped to clarify some of my mistakes.
I'm still configuring the Asterisk box - not easy at all...
Anyway, after finishing the configuration process, I will post an arranged
message to the list. I'm sure it gonna help to many users.
Eran
On 10/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Oct 17, 2007 at 11:27:59AM +0200, Ohad Levy wrote:
the PRI context
[from-pri]
X_.,1,answer
X_.,n,dial(SIP/[EMAIL PROTECTED])
This must be a copy/paste error. I figure you meant:
[from-pri]
_X.,1,answer
_X.,n,dial(SIP/[EMAIL PROTECTED])
(_X. rather than X_.). A _ at the beginning of the extension name
means it should be parsed as a pattern extension. This gives special
meaning to X, . and others.
So the above mean:
1. First-off answer the call
2. Now attempt to connect it to a specific SIP destination.
Automatically answering an incoming call may not be the best idea.
Suppose that the destination phone was busy. You still told the PRI
provider that the call was accepted, and hence the call will be charged.
Dial will answer calls when it is able to connect it with the
destination, so an explicit Answer() is not required.
Ido already gave some comments about a better usage of SIP peers rather
than mearly [EMAIL PROTECTED], so I'll assume you have a peer set up
for that destination, called dest-peer. In that case you could use:
Dial(SIP/dest-peer)
instead. This allows a better definition of codecs and such. You may
also want to dial an explicit number to the destination. For example:
Dial(SIP/dest-peer/${EXTEN})
(this dials there the number that was dialed to you, in case you have
more than one)
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il || a Mutt's
[EMAIL PROTECTED] || best
ICQ# 16849754 || friend
=
To unsubscribe, send mail to [EMAIL PROTECTED] with
the word unsubscribe in the message body, e.g., run the command
echo unsubscribe | mail [EMAIL PROTECTED]
--
Thanks,
Eran
Re: Asterisk configuration
On Wed, Oct 17, 2007 at 11:27:59AM +0200, Ohad Levy wrote:
the PRI context
[from-pri]
X_.,1,answer
X_.,n,dial(SIP/[EMAIL PROTECTED])
This must be a copy/paste error. I figure you meant:
[from-pri]
_X.,1,answer
_X.,n,dial(SIP/[EMAIL PROTECTED])
(_X. rather than X_.). A _ at the beginning of the extension name
means it should be parsed as a pattern extension. This gives special
meaning to X, . and others.
So the above mean:
1. First-off answer the call
2. Now attempt to connect it to a specific SIP destination.
Automatically answering an incoming call may not be the best idea.
Suppose that the destination phone was busy. You still told the PRI
provider that the call was accepted, and hence the call will be charged.
Dial will answer calls when it is able to connect it with the
destination, so an explicit Answer() is not required.
Ido already gave some comments about a better usage of SIP peers rather
than mearly [EMAIL PROTECTED], so I'll assume you have a peer set up
for that destination, called dest-peer. In that case you could use:
Dial(SIP/dest-peer)
instead. This allows a better definition of codecs and such. You may
also want to dial an explicit number to the destination. For example:
Dial(SIP/dest-peer/${EXTEN})
(this dials there the number that was dialed to you, in case you have
more than one)
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il || a Mutt's
[EMAIL PROTECTED] || best
ICQ# 16849754 || friend
=
To unsubscribe, send mail to [EMAIL PROTECTED] with
the word unsubscribe in the message body, e.g., run the command
echo unsubscribe | mail [EMAIL PROTECTED]
Re: Asterisk configuration
the PRI context [from-pri] X_.,1,answer X_.,n,dial(SIP/[EMAIL PROTECTED]) ( I didnt do syntax checks but thats the idea...) On 10/17/07, Eran Levy [EMAIL PROTECTED] wrote: Hi all, I've a friend which has some businesses in Canada and USA. He asked me to help him installing Asterisk on his Ubuntu 7 Linux Server. I've never did it before and this is my first time installing using Asterisk. So I started to read some tutorials and successfully installed Asterisk. Now we are moving step forward to the hardest part: the configuration… He is trying to do the following: He wants to make his Asterisk server as a gateway, which knows to receive incoming calls from a VOIP server to one of the PRIs and forward this call to a SIP server which is located in the other side… I'm searching reading a lot and I understand how to forward incoming calls to another landline or cell phone, but I can't find the answer, how to forward those calls to another SIP server… or am I missing something? Can someone help me? If you need any further information, pls let me know. Thanks! Eran
Re: Asterisk configuration
Hi Eran, On 10/17/07, Eran Levy [EMAIL PROTECTED] wrote: Hi all, I've a friend which has some businesses in Canada and USA. He asked me to help him installing Asterisk on his Ubuntu 7 Linux Server. I've never did it before and this is my first time installing using Asterisk. So I started to read some tutorials and successfully installed Asterisk. Now we are moving step forward to the hardest part: the configuration… He is trying to do the following: He wants to make his Asterisk server as a gateway, which knows to receive incoming calls from a VOIP server to one of the PRIs and forward this call to a SIP server which is located in the other side… First of all few terms you wish to know: You need to create a new SIP entry at sip.conf There are few types of SIP connections that Asterisk defines as follows: 1. peer - read only 2. user - write only 3. friend - read and write You need on both SIP servers to define something that will be able to talk to each other called inbound and outbound trunks. The definition of such trunk will be configured to know to what location to pass the call. You need to forward such request (defined at the sip.conf) to a new location. Asterisk knows how to exit such call by using a context (the right section in a dial plan). I'm searching reading a lot and I understand how to forward incoming calls to another landline or cell phone, but I can't find the answer, how to forward those calls to another SIP server… or am I missing something? Can someone help me? If you need any further information, pls let me know. Thanks! Eran I hope it points you to the right location Ido -- http://ik.homelinux.org/
