Re: [asterisk-users] sip.conf host!=dynamic peer specific options (e.g. directmedia=off, transport=tcp) not working!?
On Tue, Nov 12, 2019 at 3:06 AM Thomas Roos wrote: > Hi, > when using some non dynamic host eg. host=192.168.111.153 in sip.conf > asterisk is not considering specific peer options eg. directmedia=off, > transport=tcp > if I set host=dynamic and register the sip phone it works as expected. > Is this a bug or feature - I wanna disable the usage of directmedia for > some peers with fixed ip but wanna allow it in general. Same with > transport=tcp. > > [97] > type=peer > host=192.168.111.153 > transport=udp > context=extern > dtmfmode=auto > directmedia=off > > cheers, Thomas > Is this an incoming or outgoing call? Are you sure that entry is being used? What's the console output? -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.sangoma.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf host!=dynamic peer specific options (e.g. directmedia=off, transport=tcp) not working!?
Hi, when using some non dynamic host eg. host=192.168.111.153 in sip.conf asterisk is not considering specific peer options eg. directmedia=off, transport=tcp if I set host=dynamic and register the sip phone it works as expected. Is this a bug or feature - I wanna disable the usage of directmedia for some peers with fixed ip but wanna allow it in general. Same with transport=tcp. [97] type=peer host=192.168.111.153 transport=udp context=extern dtmfmode=auto directmedia=off cheers, Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf to pjsip.conf conversion script
On Mon, Oct 27, 2014 at 6:35 PM, John Kiniston johnkinis...@gmail.com
wrote:
Howdy,
I'm trying to get my feet wet with pjsip using the conversion script
mentioned on the Wiki on this page:
https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
I'm using the copy of the script that's included with Asterisk 13
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip
I assume I run it from /etc/asterisk with the input and output file as
arguments however there's no instructions and I don't Grok python.
Unfortunately it's not working, Despite what the below error states I do
have a udpbindaddr set to 0.0.0.0 in my configuration.
root@kiniston01:/etc/asterisk#
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py
sip.conf pjsip.conf
Traceback (most recent call last):
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 1158, in module
pjsip, non_mappings = convert(sip, pjsip_filename, dict(), False)
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 1090, in convert
map_transports(sip, pjsip, nmapped)
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 817, in map_transports
create_udp(sip, pjsip, nmapped)
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 590, in create_udp
bind = sip.multi_get('general', ['udpbindaddr', 'bindaddr'])[0]
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/astconfigparser.py,
line 407, in multi_get
(key_list, section))
LookupError: keys ['udpbindaddr', 'bindaddr'] not found for section
'general'
Based on the error I am guessing that you don't have the option
'udpbindaddr' or 'bindaddr' specified in the 'general' section of your
sip.conf. If you add one of those options to the 'general' configuration
section in your sip.conf it should hopefully work around the issue.
However, the script shouldn't error out in such a manner, so please file an
issue [1] to the bug tracker and be sure to mention the documentation too
since that should be updated as well. Asterisk issue guidelines can be
found at the following [2].
[1] https://issues.asterisk.org/jira
[2]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines#AsteriskIssueGuidelines-Submittingthebugreport
I've not turned up anything useful with Google so the mailing list is my
next step.
I can provide my configuration if needed however it is just the stock
sip.conf with a phone and two trunks added at the bottom.
Thanks!
--
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
Thanks,
--
Kevin Harwell
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] sip.conf to pjsip.conf conversion script
On Tue, Oct 28, 2014 at 9:38 AM, Kevin Harwell kharw...@digium.com wrote:
On Mon, Oct 27, 2014 at 6:35 PM, John Kiniston johnkinis...@gmail.com wrote:
Howdy,
I'm trying to get my feet wet with pjsip using the conversion script
mentioned on the Wiki on this page:
https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
I'm using the copy of the script that's included with Asterisk 13
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip
I assume I run it from /etc/asterisk with the input and output file as
arguments however there's no instructions and I don't Grok python.
Unfortunately it's not working, Despite what the below error states I do
have a udpbindaddr set to 0.0.0.0 in my configuration.
root@kiniston01:/etc/asterisk#
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py
sip.conf pjsip.conf
Traceback (most recent call last):
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 1158, in module
pjsip, non_mappings = convert(sip, pjsip_filename, dict(), False)
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 1090, in convert
map_transports(sip, pjsip, nmapped)
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 817, in map_transports
create_udp(sip, pjsip, nmapped)
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 590, in create_udp
bind = sip.multi_get('general', ['udpbindaddr', 'bindaddr'])[0]
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/astconfigparser.py,
line 407, in multi_get
(key_list, section))
LookupError: keys ['udpbindaddr', 'bindaddr'] not found for section 'general'
Based on the error I am guessing that you don't have the option 'udpbindaddr'
or 'bindaddr' specified in the 'general' section of your sip.conf. If you
add one of those options to the 'general' configuration section in your
sip.conf it should hopefully work around the issue.
However, the script shouldn't error out in such a manner, so please file an
issue [1] to the bug tracker and be sure to mention the documentation too
since that should be updated as well. Asterisk issue guidelines can be found
at the following [2].
[1] https://issues.asterisk.org/jira
[2]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines#AsteriskIssueGuidelines-Submittingthebugreport
It throws this error for me as well on the sample sip.conf, which does
have a udpbindaddr defined in the [general] context - so it's a
legitimate bug in the script.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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[asterisk-users] sip.conf to pjsip.conf conversion script
Howdy,
I'm trying to get my feet wet with pjsip using the conversion script
mentioned on the Wiki on this page:
https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
I'm using the copy of the script that's included with Asterisk 13
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip
I assume I run it from /etc/asterisk with the input and output file as
arguments however there's no instructions and I don't Grok python.
Unfortunately it's not working, Despite what the below error states I do
have a udpbindaddr set to 0.0.0.0 in my configuration.
root@kiniston01:/etc/asterisk#
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py
sip.conf pjsip.conf
Traceback (most recent call last):
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 1158, in module
pjsip, non_mappings = convert(sip, pjsip_filename, dict(), False)
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 1090, in convert
map_transports(sip, pjsip, nmapped)
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 817, in map_transports
create_udp(sip, pjsip, nmapped)
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
line 590, in create_udp
bind = sip.multi_get('general', ['udpbindaddr', 'bindaddr'])[0]
File
/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/astconfigparser.py,
line 407, in multi_get
(key_list, section))
LookupError: keys ['udpbindaddr', 'bindaddr'] not found for section
'general'
I've not turned up anything useful with Google so the mailing list is my
next step.
I can provide my configuration if needed however it is just the stock
sip.conf with a phone and two trunks added at the bottom.
Thanks!
--
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf and extension.conf configuration
Hi, The dots in extension will work as special characters. that's means in sip.conf and extensions.conf couldn't work if there's dot? thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf and extension.conf configuration
Hi, i want to ask about sip.conf extension.conf the configuration. is it possibility to make sip.conf configuration like this [1510891531543...@wlan.mnc089.mcc510.3gppnetwork.org] type = friend context = tutorial username = 1510891531543...@wlan.mnc089.mcc510.3gppnetwork.org secret = 12345 host = dynamic and the extension.conf like this exten = 1510891531557...@wlan.mnc089.mcc510.3gppnetwork.org,1,Dial(SIP/ 1510891531557...@wlan.mnc089.mcc510.3gppnetwork.org) thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf and extension.conf configuration
The dots in extension will work as special characters. On 14/09/2014 8:06 pm, rafa alfurqan rafa.alfur...@gmail.com wrote: Hi, i want to ask about sip.conf extension.conf the configuration. is it possibility to make sip.conf configuration like this [1510891531543...@wlan.mnc089.mcc510.3gppnetwork.org] type = friend context = tutorial username = 1510891531543...@wlan.mnc089.mcc510.3gppnetwork.org secret = 12345 host = dynamic and the extension.conf like this exten = 1510891531557...@wlan.mnc089.mcc510.3gppnetwork.org,1,Dial(SIP/ 1510891531557...@wlan.mnc089.mcc510.3gppnetwork.org) thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf 's tonezone option working ?
Hi, On a 11.7.0 asterisk, I'm playing with timzone option. When I'm setting this value to us or fr (as listed in indications.conf), I'm still seeing this: Language : us Tonezone : Not set Has someone a working example ? Can you reproduce this ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf and binaddr issue
10 jul 2012 kl. 20:50 skrev Kevin P. Fleming: On 07/10/2012 03:24 AM, Olle E. Johansson wrote: The Asterisk SIP channel has no knowledge about interfaces and can't bind to a specific interface for communication. In fact, it's a well known bug that if you have multiple interfaces with different IP networks, Asterisk will send from the wrong IP on some of the interfaces. Are you sure about that? The only problem area that I'm aware of is when there are multiple *overlapping* interfaces (on the same subnet, or providing the same route(s)). In that case, Asterisk can receive messages on one IP address out of the overlapping set, but reply using a different one from the set, because it doesn't specify the source IP address and instead lets the UDP/IP stack select one. If the interfaces don't overlap in any way, I don't see how it would be possible for Asterisk to send messages with the wrong source IP address, since it does not specify the source IP address at all. If this is occurring, it must involve the operating system's IP stack in some fashion. Yes, I still use quite a lot of IPtables tricks to overcome this issue. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf and binaddr issue
6 jul 2012 kl. 23:18 skrev Felix Salfelder: Hi there. i am seriously stuck with an asterisk and sip problem. the following sip.conf works with respect to some_peer: [general] bindaddr = x.y.z.w nat = no [some_peer] type=peer host=somehost secret=somesecret some other unrelated options here x.y.z.w is the ip address of the interface pointing to the network containing somehost. more precisely its the address of tun0 and route -n prints Destination Gateway Genmask Flags Metric RefUse Iface [..] x.y.z.0 0.0.0.0 255.255.255.0 U 0 0 0 tun0 [..] here 'it works' implies that i have to change and reload sip.conf after ifup tun0, or anything that forces tun0 to go down, like my dsl provider. also, the bindaddr line is suboptimal for the other peers... the same thing -- without the bindaddr part -- doesnt work. more precisely it almost works. its just incoming sound that doesnt. this must have something to do with how asterisk picks up interface addresses and communicates them to the peer in question. inspecting the packages sent to somehost, gave me the impression that asterisk uses the ip adress of ppp0 (a dsl modem) instead. how am i supposed to tell asterisk to use tun0 as the interface for [some_peer] so i can remove the bindaddr line? i've found many nat-related options in the manual, but there is no nat involved here. also, i couldnt find anything similar to iface=tun0, although the sip dialogue apparently relies on ip adresses and routing. this is about asterisk on sid, version 1:1.8.13.0~dfsg-1, but of course i'm going to switch to whatever you might suggest. The Asterisk SIP channel has no knowledge about interfaces and can't bind to a specific interface for communication. In fact, it's a well known bug that if you have multiple interfaces with different IP networks, Asterisk will send from the wrong IP on some of the interfaces. Sorry, /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf and bindaddr issue
On Tue, Jul 10, 2012 at 10:24:18AM +0200, Olle E. Johansson wrote: The Asterisk SIP channel has no knowledge about interfaces and can't bind to a specific interface for communication. Thanks for the reply. in the meantime i've found a sort of workaround. [general] host = dynamic ; take some local, static address bindaddr = 192.168.1.1 ; and don't use that address very much localnet = 192.168.0.0/255.255.0.0 ; ... [sip_out] ; pretend nat nat = route ; ... i'm not sure about all implications. for example, incoming connections must be handled with iptables, and in the first second of a call (from sip_out) theres no sound. i can live with that for a while. In fact, it's a well known bug that if you have multiple interfaces with different IP networks, Asterisk will send from the wrong IP on some of the interfaces. couldn't find much about it on the net. anyway, if it's well known: what would be the downside of just (silently, implicitly) taking the right adress? like when when resolving the peer-host, take a look into the routing table...? regards felix -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf and binaddr issue
On 07/10/2012 03:24 AM, Olle E. Johansson wrote: The Asterisk SIP channel has no knowledge about interfaces and can't bind to a specific interface for communication. In fact, it's a well known bug that if you have multiple interfaces with different IP networks, Asterisk will send from the wrong IP on some of the interfaces. Are you sure about that? The only problem area that I'm aware of is when there are multiple *overlapping* interfaces (on the same subnet, or providing the same route(s)). In that case, Asterisk can receive messages on one IP address out of the overlapping set, but reply using a different one from the set, because it doesn't specify the source IP address and instead lets the UDP/IP stack select one. If the interfaces don't overlap in any way, I don't see how it would be possible for Asterisk to send messages with the wrong source IP address, since it does not specify the source IP address at all. If this is occurring, it must involve the operating system's IP stack in some fashion. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf and binaddr issue
Hi there. i am seriously stuck with an asterisk and sip problem. the following sip.conf works with respect to some_peer: [general] bindaddr = x.y.z.w nat = no [some_peer] type=peer host=somehost secret=somesecret some other unrelated options here x.y.z.w is the ip address of the interface pointing to the network containing somehost. more precisely its the address of tun0 and route -n prints Destination Gateway Genmask Flags Metric RefUse Iface [..] x.y.z.0 0.0.0.0 255.255.255.0 U 0 0 0 tun0 [..] here 'it works' implies that i have to change and reload sip.conf after ifup tun0, or anything that forces tun0 to go down, like my dsl provider. also, the bindaddr line is suboptimal for the other peers... the same thing -- without the bindaddr part -- doesnt work. more precisely it almost works. its just incoming sound that doesnt. this must have something to do with how asterisk picks up interface addresses and communicates them to the peer in question. inspecting the packages sent to somehost, gave me the impression that asterisk uses the ip adress of ppp0 (a dsl modem) instead. how am i supposed to tell asterisk to use tun0 as the interface for [some_peer] so i can remove the bindaddr line? i've found many nat-related options in the manual, but there is no nat involved here. also, i couldnt find anything similar to iface=tun0, although the sip dialogue apparently relies on ip adresses and routing. this is about asterisk on sid, version 1:1.8.13.0~dfsg-1, but of course i'm going to switch to whatever you might suggest. regards and thanks felix -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip.conf and extensions.conf configuration for Exchange 2010 U.M.
Hi All, I'm using Exchange as our voicemail system. Everything works fine until the 1 week mark when Exchange changes the port number used, then Asterisk 1.8 seg faults and I have no phones (unless I restart the U.M. service before the 1 week period is up). Since that is a hack, I'm hoping someone can post their working configs that accomodates the port change. The documentation I've seen is still a little unclear to me. I'm not using secured mode, so just using ports 5065/5067. Thanks for your help. Jim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf, realtime, and LDAP
On Mon, Dec 27, 2010 at 3:38 AM, Olivier oza_4...@yahoo.fr wrote: 2010/12/26 Richard Kenner ken...@gnat.com I'm confused exactly what's supported with LDAP and Asterisk. What I want to do is to have SIP peer information read directly (in realtime) from LDAP. Can this be done? If so, with what Asterisk versions? I'm also a bit confused about what's possible and not possible. Anyway, my understanding is : - you can directly query an LDAP directory from your dialplan (LDAPget), - you can also use Asterisk Realtime Architecture and use LDAP as a backend. It can be used with any Asterisk version (at least 1.4 and later). Cheers http://www.zentyal.org/ does a really good job with the LDAP integration. Try it out and see what they did. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf, realtime, and LDAP
2010/12/26 Richard Kenner ken...@gnat.com I'm confused exactly what's supported with LDAP and Asterisk. What I want to do is to have SIP peer information read directly (in realtime) from LDAP. Can this be done? If so, with what Asterisk versions? I'm also a bit confused about what's possible and not possible. Anyway, my understanding is : - you can directly query an LDAP directory from your dialplan (LDAPget), - you can also use Asterisk Realtime Architecture and use LDAP as a backend. It can be used with any Asterisk version (at least 1.4 and later). Cheers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf, realtime, and LDAP
I'm confused exactly what's supported with LDAP and Asterisk. What I want to do is to have SIP peer information read directly (in realtime) from LDAP. Can this be done? If so, with what Asterisk versions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf, realtime, and LDAP
On Sat, Dec 25, 2010 at 8:58 PM, Richard Kenner ken...@gnat.com wrote: I'm confused exactly what's supported with LDAP and Asterisk. What I want to do is to have SIP peer information read directly (in realtime) from LDAP. Can this be done? If so, with what Asterisk versions? -- Here is the link https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver Test and add comments to help the documentation grow. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf register in realtime DB
On 08/07/2010 01:11 AM, unsero...@aol.com wrote: Why don't you use 'real' realtime meaning to have your sip peers in your database? Then you would not have to do a reload after adding new peers to your db. And you can still have sip peers additionally in sip.conf. I have all of my sip peers in a realtime database, that is not the question ! The question is that I want the register-statements of sip.conf in a realtime database. I've posted all the info to do so in my first post. But it conflicts for some unknown reason with realtime SIP peers. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf register in realtime DB
Please can anyone help me with this ?! I have tried renaming the sip.conf file, or tried including another file into sip.conf like sippy.conf and then add sippy.conf = mysql,AsteriskDB,ast_config to extconfig.conf but all this is not working. The only thing that changes something is my example listed below, but then I always have that nasty WARNING which I find odd. I need realtime sip registrations (so without having to do a sip reload). Kind regards, Jonas. On 08/03/2010 10:13 AM, Jonas Kellens wrote: Hello list, scrambling different pieces of info together I've come with the following : I want to have my register = statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:passw...@sip.provider.net In ext_config (text file) I have : sipusers = mysql,AsteriskDB,sip_buddies sippeers = mysql,AsteriskDB,sip_buddies sip.conf = mysql,AsteriskDB,ast_config In sip.conf (text file) I have also : sip.conf : rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis? (yes|no) After a reload I noticed that the registration came through when I executed sip show registrations. This realtime works. But I then get a lot of the following messages : / [Aug 2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'user2' [Aug 2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'user2'/ rtcachefriends is turned on (see above) qualify is on on every peer (and I want it to stay that way) Can anyone tell me what I need to configure to get a 100% working example ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf register in realtime DB
You cannot use realtime static and the other realtime tables at the same time. You will need to use realtime and then use something like the EXEC command in sip.conf to execute a script that then pulls the register statement from your database. Or use the realtime static table for everything. On Fri, 2010-08-06 at 10:57 +0200, Jonas Kellens wrote: Please can anyone help me with this ?! I have tried renaming the sip.conf file, or tried including another file into sip.conf like sippy.conf and then add sippy.conf = mysql,AsteriskDB,ast_config to extconfig.conf but all this is not working. The only thing that changes something is my example listed below, but then I always have that nasty WARNING which I find odd. I need realtime sip registrations (so without having to do a sip reload). Kind regards, Jonas. On 08/03/2010 10:13 AM, Jonas Kellens wrote: Hello list, scrambling different pieces of info together I've come with the following : I want to have my register = statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:passw...@sip.provider.net In ext_config (text file) I have : sipusers = mysql,AsteriskDB,sip_buddies sippeers = mysql,AsteriskDB,sip_buddies sip.conf = mysql,AsteriskDB,ast_config In sip.conf (text file) I have also : sip.conf : rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis? (yes|no) After a reload I noticed that the registration came through when I executed sip show registrations. This realtime works. But I then get a lot of the following messages : [Aug 2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'user2' [Aug 2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'user2' rtcachefriends is turned on (see above) qualify is on on every peer (and I want it to stay that way) Can anyone tell me what I need to configure to get a 100% working example ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf register in realtime DB
On 08/06/2010 06:45 PM, Carlos Chavez wrote: You cannot use realtime static and the other realtime tables at the same time. You will need to use realtime and then use something like the EXEC command in sip.conf to execute a script that then pulls the register statement from your database. Or use the realtime static table for everything. Using the EXEC-command in sip.conf means I will have to issue a sip reload when I want to load changes in the database ?! New information that is put into the REGISTER-database is not available without a 'sip reload' ?! If not, do you see another way to have new registrations without a 'sip reload' ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf register in realtime DB
On 08/06/2010 06:45 PM, Carlos Chavez wrote: Or use the realtime static table for everything. What do you mean by everything ?! What is this everything ?! You mean all the sip options in a database and so no sip.conf file ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf register in realtime DB
You cannot use realtime static and the other realtime tables at the same time. You will need to use realtime and then use something like the EXEC command in sip.conf to execute a script that then pulls the register statement from your database. Or use the realtime static table for everything. Using the EXEC-command in sip.conf means I will have to issue a sip reload when I want to load changes in the database ?! New information that is put into the REGISTER-database is not available without a 'sip reload' ?! If not, do you see another way to have new registrations without a 'sip reload' ?! Kind regards, Jonas. -- Why don't you use 'real' realtime meaning to have your sip peers in your database? Then you would not have to do a reload after adding new peers to your db. And you can still have sip peers additionally in sip.conf. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf register in realtime DB
Hello list, scrambling different pieces of info together I've come with the following : I want to have my register = statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:passw...@sip.provider.net In ext_config (text file) I have : sipusers = mysql,AsteriskDB,sip_buddies sippeers = mysql,AsteriskDB,sip_buddies sip.conf = mysql,AsteriskDB,ast_config In sip.conf (text file) I have also : sip.conf : rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis? (yes|no) After a reload I noticed that the registration came through when I executed sip show registrations. This realtime works. But I then get a lot of the following messages : / [Aug 2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'user2' [Aug 2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'user2'/ rtcachefriends is turned on (see above) qualify is on on every peer (and I want it to stay that way) Can anyone tell me what I need to configure to get a 100% working example ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf User vs Username
Hi In sip.conf, you generally have something like [name] .. username= secret= What is the difference between the name specified in brackets and the username key ? What the sip client should provide ? What do we use in dialplan when trying to reach this client ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf User vs Username
On Tue, Jul 6, 2010 at 7:11 PM, Ruddy Gbaguidi plugwo...@micnes.com wrote: What is the difference between the name specified in brackets and the username key ? Context and username. What the sip client should provide ? The client will tell you their settings What do we use in dialplan when trying to reach this client ? Dial(SIP/Context) This is all documented in sip.conf, otherwise the book (http://astbook.asteriskdocs.org/). -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf - sort order, does it matter
On 02/19/10 08:54, Olle E. Johansson wrote: 17 feb 2010 kl. 19.12 skrev Joseph: Does the sort order matter in sip.conf file? I know sort order might effect: allow=ulaw allow=alaw but does it matter where I place: insecure=invite ? The reason I'm asking is that I've loaded almost two identical (sip.conf and extension.conf) files on the same asterisk server and with one set insecure=invite is working correctly. When I load the second set of dial plan (sip.conf and extension.conf) insecure=invite is not taking effect. I get: ... username mismatch, have 4, digest has pstn- handle_request_invite: Failed to authenticate user KMIEC J You propably have a type=friend where the user part matches before you even hit the peer part, where the insecure configuration parameter matches. There is a confusion here on the From: username and the authentication username used, so there is a challenge sent. /O Yes, I have type=friend but I've loaded my other dial plan where I have type=friend as well and insecure=invite is working. So, it must be the sort order that is generating problems. -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf - sort order, does it matter
17 feb 2010 kl. 19.12 skrev Joseph: Does the sort order matter in sip.conf file? I know sort order might effect: allow=ulaw allow=alaw but does it matter where I place: insecure=invite ? The reason I'm asking is that I've loaded almost two identical (sip.conf and extension.conf) files on the same asterisk server and with one set insecure=invite is working correctly. When I load the second set of dial plan (sip.conf and extension.conf) insecure=invite is not taking effect. I get: ... username mismatch, have 4, digest has pstn- handle_request_invite: Failed to authenticate user KMIEC J You propably have a type=friend where the user part matches before you even hit the peer part, where the insecure configuration parameter matches. There is a confusion here on the From: username and the authentication username used, so there is a challenge sent. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf - sort order, does it matter
On Fri, Feb 19, 2010 at 8:54 AM, Olle E. Johansson o...@edvina.net wrote: You propably have a type=friend where the user part matches before you even hit the peer part, where the insecure configuration parameter matches. There is a confusion here on the From: username and the authentication username used, so there is a challenge sent. Is it just me, or would it be nice if a clear, understandable and unambiguous way to express codec desirata was invented? Is there a future iteration of SIP that deals with it? /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf - sort order, does it matter
19 feb 2010 kl. 10.22 skrev Randy R: On Fri, Feb 19, 2010 at 8:54 AM, Olle E. Johansson o...@edvina.net wrote: You propably have a type=friend where the user part matches before you even hit the peer part, where the insecure configuration parameter matches. There is a confusion here on the From: username and the authentication username used, so there is a challenge sent. Is it just me, or would it be nice if a clear, understandable and unambiguous way to express codec desirata was invented? Is there a future iteration of SIP that deals with it? It's not only SIP, it's the whole Asterisk codec negotiation framework that needs a serious overhaul: Please read: http://www.voip-forum.com/asterisk/2010-02/asterisk-18-lts-wishlist-1-media-negiotiation-framework/ Interestingly enough, this blog post (and the same message on asterisk-dev) has got NO feedback, even though this has been a hot topic for years. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf - sort order, does it matter
17 feb 2010 kl. 19.12 skrev Joseph: Does the sort order matter in sip.conf file? I know sort order might effect: allow=ulaw allow=alaw but does it matter where I place: insecure=invite ? The reason I'm asking is that I've loaded almost two identical (sip.conf and extension.conf) files on the same asterisk server and with one set insecure=invite is working correctly. When I load the second set of dial plan (sip.conf and extension.conf) insecure=invite is not taking effect. I get: ... username mismatch, have 4, digest has pstn- handle_request_invite: Failed to authenticate user KMIEC J Someone have mentioned that sort order in sip.conf might effect the way it works. No, where you place the insecure=invite in the device specification does not matter. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf - sort order, does it matter
On 02/18/10 09:00, Olle E. Johansson wrote: 17 feb 2010 kl. 19.12 skrev Joseph: Does the sort order matter in sip.conf file? I know sort order might effect: allow=ulaw allow=alaw but does it matter where I place: insecure=invite ? The reason I'm asking is that I've loaded almost two identical (sip.conf and extension.conf) files on the same asterisk server and with one set insecure=invite is working correctly. When I load the second set of dial plan (sip.conf and extension.conf) insecure=invite is not taking effect. I get: ... username mismatch, have 4, digest has pstn- handle_request_invite: Failed to authenticate user KMIEC J Someone have mentioned that sort order in sip.conf might effect the way it works. No, where you place the insecure=invite in the device specification does not matter. /O But for some reason or the other it insecure=invite works in one sip.conf but not the other; that is what puzzling me. I've compered two sip.conf with meld and they are identical except some registration context and numbers. -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf - sort order, does it matter
Does the sort order matter in sip.conf file? I know sort order might effect: allow=ulaw allow=alaw but does it matter where I place: insecure=invite ? The reason I'm asking is that I've loaded almost two identical (sip.conf and extension.conf) files on the same asterisk server and with one set insecure=invite is working correctly. When I load the second set of dial plan (sip.conf and extension.conf) insecure=invite is not taking effect. I get: ... username mismatch, have 4, digest has pstn- handle_request_invite: Failed to authenticate user KMIEC J Someone have mentioned that sort order in sip.conf might effect the way it works. -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem
As a guess, they can both talk to the server, but can't talk to each other. What is common to that is they may be trying to reinvite each other, and there is no path through the respective routers/firewalls to the other. So if reinvite is set to yes, set it to no, in both phone profiles on the server. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves Arikoglu Sent: Monday, January 25, 2010 7:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y eth0 of the server is configured to 10.26.192.107 The Problem: SIP registration works, phone rings in- and outbound, but there is no audio, nor the caller neither the callee can hear anything. So i am quite sure that is has something to do with firewalls, natting and so on but i?ve read hundreds of pages and tried thousands of setting but i cant get audio to work.. the strange thing is... when i call the versatel-sip-number from my mobile phone, i see the call coming in in the cli, i see the voiceprompts that asterisk plays, but even there I cant hear anything on my mobile. next strange thing: i defined 2 sip-extensions. both are registered... everything is fine... routes are ok, they can call out and can be called from external and from internal (sip phones call each other).. but the same... no audio. but when one sip extension calls a wrong number... the cannot be completed message is hearable. i configured a queue with moh and even this works... but why cant to sip-phones talk to each other? why cant an external caller hear any audio? if i make sip debug, i see traffic (and due to extension is calling i think that on the sip-level everything is okay...) how can i see, which port and interface is chosen for audio when a call comes in? thanks, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf with versatel and two NICs very strange problem
Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y eth0 of the server is configured to 10.26.192.107 The Problem: SIP registration works, phone rings in- and outbound, but there is no audio, nor the caller neither the callee can hear anything. So i am quite sure that is has something to do with firewalls, natting and so on but i?ve read hundreds of pages and tried thousands of setting but i cant get audio to work.. the strange thing is... when i call the versatel-sip-number from my mobile phone, i see the call coming in in the cli, i see the voiceprompts that asterisk plays, but even there I cant hear anything on my mobile. next strange thing: i defined 2 sip-extensions. both are registered... everything is fine... routes are ok, they can call out and can be called from external and from internal (sip phones call each other).. but the same... no audio. but when one sip extension calls a wrong number... the cannot be completed message is hearable. i configured a queue with moh and even this works... but why cant to sip-phones talk to each other? why cant an external caller hear any audio? if i make sip debug, i see traffic (and due to extension is calling i think that on the sip-level everything is okay...) how can i see, which port and interface is chosen for audio when a call comes in? thanks, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem
thanks, i tried this already but unfortunately no change. any further suggestions or answers concerning my other questions? thanx, yves Cary Fitch schrieb: As a guess, they can both talk to the server, but can't talk to each other. What is common to that is they may be trying to reinvite each other, and there is no path through the respective routers/firewalls to the other. So if reinvite is set to yes, set it to no, in both phone profiles on the server. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves Arikoglu Sent: Monday, January 25, 2010 7:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y eth0 of the server is configured to 10.26.192.107 The Problem: SIP registration works, phone rings in- and outbound, but there is no audio, nor the caller neither the callee can hear anything. So i am quite sure that is has something to do with firewalls, natting and so on but i?ve read hundreds of pages and tried thousands of setting but i cant get audio to work.. the strange thing is... when i call the versatel-sip-number from my mobile phone, i see the call coming in in the cli, i see the voiceprompts that asterisk plays, but even there I cant hear anything on my mobile. next strange thing: i defined 2 sip-extensions. both are registered... everything is fine... routes are ok, they can call out and can be called from external and from internal (sip phones call each other).. but the same... no audio. but when one sip extension calls a wrong number... the cannot be completed message is hearable. i configured a queue with moh and even this works... but why cant to sip-phones talk to each other? why cant an external caller hear any audio? if i make sip debug, i see traffic (and due to extension is calling i think that on the sip-level everything is okay...) how can i see, which port and interface is chosen for audio when a call comes in? thanks, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf with versatel and two NICs very strange problem
- Yves Arikoglu yves...@gmx.de wrote: Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y Either a typo or you have an IP conflict? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf with versatel and two NICs very strange problem
thanx... a typo... the routers local ip is 10.26.208.253 yves Tim Nelson schrieb: - Yves Arikoglu yves...@gmx.de wrote: Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y Either a typo or you have an IP conflict? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf parameter and sip msg between server - client
- what's the difference between a subscribe request et a register request ? A subscription in the SIP protocol is saying Hey, I'd like to be notified when something happens. This is most often used when a phone wants to subscribe to the state of another extension, or to the status of a voicemail box. A registration is where one SIP device tells another Hey, I'm over here. If you get any calls for me, send them to me at this IP address and port. -- Jared Smith Training Manager Digium, Inc. Hi Jared Thx for your explanation. So in sip.conf if I set allowsubscribe to no in [general] section that will mean by default all my device I will declare will no be able to receive SUBSCRIPTION msg ? Except if I declare allowsubscribe for one of them... now I have a better understanding of what a subscription is, these questions are coming about some parameter in sip.conf : - 'notifyringing' set to 'yes' will send sorry but I'm already on call to any device when my device named john is already in use ? Is it an issue to limit incoming/outgoing call for a device ? - 'callcounter' is it an issue to limit incoming/outgoing call for a device ? - I read that 'call-limit' or 'busylevel' (in asterisk 1.6) is an issue to limit incoming/outgoing call on a device. Is it true ? Can I use them together or do I have to use just one ? Regards Harry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf parameter and sip msg between server - client
Hello I have few questions : - what's the difference between a subscribe request et a register request ? - in asterisk 1.6 allowguest=yes or no param does it work ? if yes, please someone could explain how doest it work because I think i'm a little bit confuse. - if I configure a sip terminal in sip.conf like this [john] type=friend username=JOHN secret=mypassword host=dynamic context=default these affirmations are right or wrong : a) 'john' and 'mypassword' are variables which are used when I want to connect my softphone or phone to Asterisk server (register request) AND when I initiate a call (invite request)? b) 'dynamic' mean that [john] will be automatically registred to Asterisk server and 'qualify=yes' parameter may not be necessary ? c) in your softphone setting (here i use xlite), parameter 'username' must be the same as parameter 'username' in sip.conf ? d) in you softphone setting (here i use xlite), parameter 'Authorization username' must be the same as parameter [john] in sip.conf ? e) instead of using 'dynamic' for parameter 'host', if I use @IP or a hostname or FQDN, parameter qualify must set to' yes' or to '2000' in order to Asterisk server can know when [john] is reachable ? regards H. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf parameter and sip msg between server - client
Hello, well let me explain one part of your question, the host parameter. if you want to restrict the access to one ip you can say it here. host=192.168.2.13 means, that you can only use this account from 192.168.0.13, eg. for security reasons. i recommend so set it to dynamic at the moment and if it works you can change if you wish. aot of information about the sip.conf you can find here: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf On Wed, Aug 5, 2009 at 2:32 PM, harry Rrhm.noa...@gmail.com wrote: Hello I have few questions : - what's the difference between a subscribe request et a register request ? - in asterisk 1.6 allowguest=yes or no param does it work ? if yes, please someone could explain how doest it work because I think i'm a little bit confuse. - if I configure a sip terminal in sip.conf like this [john] type=friend username=JOHN secret=mypassword host=dynamic context=default these affirmations are right or wrong : a) 'john' and 'mypassword' are variables which are used when I want to connect my softphone or phone to Asterisk server (register request) AND when I initiate a call (invite request)? b) 'dynamic' mean that [john] will be automatically registred to Asterisk server and 'qualify=yes' parameter may not be necessary ? c) in your softphone setting (here i use xlite), parameter 'username' must be the same as parameter 'username' in sip.conf ? d) in you softphone setting (here i use xlite), parameter 'Authorization username' must be the same as parameter [john] in sip.conf ? e) instead of using 'dynamic' for parameter 'host', if I use @IP or a hostname or FQDN, parameter qualify must set to' yes' or to '2000' in order to Asterisk server can know when [john] is reachable ? regards H. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf parameter and sip msg between server - client
On Wed, 2009-08-05 at 14:32 +0200, harry R wrote: - what's the difference between a subscribe request et a register request ? A subscription in the SIP protocol is saying Hey, I'd like to be notified when something happens. This is most often used when a phone wants to subscribe to the state of another extension, or to the status of a voicemail box. A registration is where one SIP device tells another Hey, I'm over here. If you get any calls for me, send them to me at this IP address and port. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf RTP settings
I have the following set in sip.conf [general] section. rtptimeout = 60 rtpholdtimeout = 300 I would like to set these to default, or null the general settings for one upline friend as it is solely a fax peer (T38 over SIP) How can this be easily done? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf outboundproxy
John Todd wrote: Would it be so difficult to have perhaps two different proxies? One would be for any SIP messages destined for IP addresses that were not in any of the localnet= lines, and one would be for any SIP messages destined for IP addresses that were destined for IP addresses that were NOT in the localnet= lines. Of course, leaving them blank would mean that a proxy would not be used for one group or the other. This would allow creation of the concept of outside and inside at an administrative level using previously-described network definitions in sip.conf. Plus, it would dis-entangle a lot of the logic that one might otherwise have to install on the proxy to reflect certain messages back into NATted zones or otherwise complex internal structures. I don't think this is the right distinction; really, you have a list of 'known' hosts that you don't need to go through the proxy to reach, and you go through the proxy to reach the 'unknown' hosts. And, in Asterisk 1.6.x, you can already set the outboundproxy setting at the general level and on a per-peer basis. So, for all your phones/internal servers/etc., just set them to not use the proxy. In fact, this is even better when one of your 'internal' phones happens to be registered from a non-'localnet' IP address. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf outboundproxy
So, does anyone ever used outboundproxy in sip.conf with success? Does it only send OUTBOUND calls via the proxy and not also internal extension calls via that proxy? Best Regards, Ricardo. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf outboundproxy
Ricardo Carvalho wrote: Does it only send OUTBOUND calls via the proxy and not also internal extension calls via that proxy? As has already been posted in your other threads about this subject, Asterisk has no concept of an 'outbound' call at all. In that sense, the name of this option in sip.conf is incorrect, it should just be 'proxy'. If you tell Asterisk to use a SIP proxy for sending out SIP requests, it will send all requests to that proxy, regardless of whether that request might be involved in a call that you classify as 'internal'. To Asterisk, a SIP call is a SIP call; there is no 'internal', 'external', 'outbound', 'inbound', at least not in the sense of 'inside my PBX' or 'outside my PBX'. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf outboundproxy
Thanks Kevin. Although it doesn't fit my needs, thanks for the explanation. I guess I'll really have to combine Asterisk with OpenSer to do what I want. Ricardo. On Thu, Mar 26, 2009 at 1:07 PM, Kevin P. Fleming kpflem...@digium.comwrote: Ricardo Carvalho wrote: Does it only send OUTBOUND calls via the proxy and not also internal extension calls via that proxy? As has already been posted in your other threads about this subject, Asterisk has no concept of an 'outbound' call at all. In that sense, the name of this option in sip.conf is incorrect, it should just be 'proxy'. If you tell Asterisk to use a SIP proxy for sending out SIP requests, it will send all requests to that proxy, regardless of whether that request might be involved in a call that you classify as 'internal'. To Asterisk, a SIP call is a SIP call; there is no 'internal', 'external', 'outbound', 'inbound', at least not in the sense of 'inside my PBX' or 'outside my PBX'. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf outboundproxy
On Mar 26, 2009, at 6:07 AM, Kevin P. Fleming wrote: Ricardo Carvalho wrote: Does it only send OUTBOUND calls via the proxy and not also internal extension calls via that proxy? As has already been posted in your other threads about this subject, Asterisk has no concept of an 'outbound' call at all. In that sense, the name of this option in sip.conf is incorrect, it should just be 'proxy'. If you tell Asterisk to use a SIP proxy for sending out SIP requests, it will send all requests to that proxy, regardless of whether that request might be involved in a call that you classify as 'internal'. To Asterisk, a SIP call is a SIP call; there is no 'internal', 'external', 'outbound', 'inbound', at least not in the sense of 'inside my PBX' or 'outside my PBX'. I agree. But... (isn't there always a caveat?) Would it be so difficult to have perhaps two different proxies? One would be for any SIP messages destined for IP addresses that were not in any of the localnet= lines, and one would be for any SIP messages destined for IP addresses that were destined for IP addresses that were NOT in the localnet= lines. Of course, leaving them blank would mean that a proxy would not be used for one group or the other. This would allow creation of the concept of outside and inside at an administrative level using previously-described network definitions in sip.conf. Plus, it would dis-entangle a lot of the logic that one might otherwise have to install on the proxy to reflect certain messages back into NATted zones or otherwise complex internal structures. I have imagined several more complex situations where I'd want to have multiple proxies, each with their own network ACL trigger masks, but I'll stick with the simple case for now. :-) JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf outboundproxy
The problem is that I cannot put the outboundproxy statement to the applicable sip extension context, due to the fact that I want to force every ENUM call to go via the proxy; and ENUM calls don't use any context to leave asterisk. Even so, putting outboundproxy statement is in the global section of sip.conf, for internal calls destined to phones registered in the same asterisk server, I think asterisk should see those are internal calls and don't ship the signaling through the proxy, right? Ricardo. On Tue, Mar 24, 2009 at 7:08 PM, Danny Nicholas da...@debsinc.com wrote: Just a guess, but your outboundproxy statement is in the global section of sip.conf, which is making it apply to all sip traffic. If you move that line to the applicable sip extension (ie. prox...@sipprov.com), this will probably fix the behavior, even if it doesn't resolve the problem. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Tuesday, March 24, 2009 1:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip.conf outboundproxy On 24 Mar 2009, at 17:51, Ricardo Carvalho wrote: Hi, I'm trying to enable sip.conf outboundproxy support in version 1.4.20.1 of Asterisk, but for the tests I made, every calls, even internal SIP calls between extensions are sent over the proxy that I have specified with the outboundproxy=xxx.xxx.xxx.xxx in sip.conf. I think this isn't the expected behaviour, right? Only OUTBOUND calls should go through the proxy, right? Never used it before, but in the mind of Asterisk, how is your sip handset any different to a provider? Its outbound from asterisk.. I may be wrong.. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf outboundproxy
Hi, I'm trying to enable sip.conf outboundproxy support in version 1.4.20.1 of Asterisk, but for the tests I made, every calls, even internal SIP calls between extensions are sent over the proxy that I have specified with the outboundproxy=xxx.xxx.xxx.xxx in sip.conf. I think this isn't the expected behaviour, right? Only OUTBOUND calls should go through the proxy, right? Am I doing something wrong or is this the real behaviour of the outboundproxy variable in sip.conf? Best Regards, Ricardo Carvalho. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf outboundproxy
On 24 Mar 2009, at 17:51, Ricardo Carvalho wrote: Hi, I'm trying to enable sip.conf outboundproxy support in version 1.4.20.1 of Asterisk, but for the tests I made, every calls, even internal SIP calls between extensions are sent over the proxy that I have specified with the outboundproxy=xxx.xxx.xxx.xxx in sip.conf. I think this isn't the expected behaviour, right? Only OUTBOUND calls should go through the proxy, right? Never used it before, but in the mind of Asterisk, how is your sip handset any different to a provider? Its outbound from asterisk.. I may be wrong.. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf outboundproxy
Just a guess, but your outboundproxy statement is in the global section of sip.conf, which is making it apply to all sip traffic. If you move that line to the applicable sip extension (ie. prox...@sipprov.com), this will probably fix the behavior, even if it doesn't resolve the problem. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Tuesday, March 24, 2009 1:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip.conf outboundproxy On 24 Mar 2009, at 17:51, Ricardo Carvalho wrote: Hi, I'm trying to enable sip.conf outboundproxy support in version 1.4.20.1 of Asterisk, but for the tests I made, every calls, even internal SIP calls between extensions are sent over the proxy that I have specified with the outboundproxy=xxx.xxx.xxx.xxx in sip.conf. I think this isn't the expected behaviour, right? Only OUTBOUND calls should go through the proxy, right? Never used it before, but in the mind of Asterisk, how is your sip handset any different to a provider? Its outbound from asterisk.. I may be wrong.. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP.Conf - bindaddr per peer?
31 jan 2009 kl. 02.44 skrev Mike: Replying to my own message. How difficult would it be to add a bindaddr (and possibly bindport) PER PEER in SIP.conf? How much of a bounty would I have to pay to get this done you think? Well, if you run bindaddr=0.0.0.0 Asterisk will listen to all IP's. I would say the simplest way would be to implement some sort of ACL for which address a peer accept inbound communication. The problem here is making sure that we send From the proper IP. It can be done, but with testing it's propably a couple of days work. Adding bindport would be a huge project, since it requires multiple ports in parallell, something that we're still a bit nervous about doing in chan_sip for 1.6 with the addition of TLS and TCP. The SIP structure locking scheme is... Well, to put it mildly, scary. For pricing, I would suggest you use the -biz list or send private e- mails. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP.Conf - bindaddr per peer?
hI, Trying to understand how to setup two PRIs in sip.conf. Using Asterisk 1.4.23. I have a provider giving me two PRI (different rate centers) through SIP. Both PRI comes in from the same IP on the provider side, but go to two different IPs (both on the same box) on my side. How can I setup two different SIP peer, one for each of the PRIs I get, if all I can use to differenciate them are the IP address ? I can't find any obvious setting in the sip.conf peer settings. The general section has bindaddr which would make sense, but since it's general and not per peer it's of no use Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP.Conf - bindaddr per peer?
30 jan 2009 kl. 16.59 skrev Mike: hI, Trying to understand how to setup two PRIs in sip.conf. Using Asterisk 1.4.23. I have a provider giving me two PRI (different rate centers) through SIP. Both PRI comes in from the same IP on the provider side, but go to two different IPs (both on the same box) on my side. How can I setup two different SIP peer, one for each of the PRIs I get, if all I can use to differenciate them are the IP address…? I can't find any obvious setting in the sip.conf peer settings. The general section has bindaddr which would make sense, but since it's general and not per peer it's of no use… Interesting question. I don't think you can. The ACLs only work on sender's address. Maybe we could consider at some point implement local ACLs too. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP.Conf - bindaddr per peer?
Replying to my own message. How difficult would it be to add a bindaddr (and possibly bindport) PER PEER in SIP.conf? How much of a bounty would I have to pay to get this done you think? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, January 30, 2009 10:59 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] SIP.Conf - bindaddr per peer? hI, Trying to understand how to setup two PRIs in sip.conf. Using Asterisk 1.4.23. I have a provider giving me two PRI (different rate centers) through SIP. Both PRI comes in from the same IP on the provider side, but go to two different IPs (both on the same box) on my side. How can I setup two different SIP peer, one for each of the PRIs I get, if all I can use to differenciate them are the IP address ? I can't find any obvious setting in the sip.conf peer settings. The general section has bindaddr which would make sense, but since it's general and not per peer it's of no use Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf templates and realtime
I currently have my phones setup in the sip.conf file. I use templates to describe the specific settings to each phone type. For instance in sip.conf, I have: [generic_phone](!) ... ... [polycom501](!,generic_phone) ... ... [grandstream](!,generic_phone) ... ... ;begin subscribers [200](polycom501) ... ... [201](grandstream) ... ... I am using asterisk 1.4.21.2 I would like to move my sip users to realtime, so my questions are: 1) Can I continue to use the templates from sip.conf and the template settings get passed to realtime and if so, how? In the comments in the sip.conf file where it shows the User config options ant Peer configuration, on the peer side it shows a template field, which seems to indicate to me that this can be done. 2) If this is not the purpose of the template field, what is it's purpose? I can not seem to find it documented anywhere. Note: I do not have any problems getting realtime to work, as long as I put every field that is needed (or required) in each record, but I think life would be easier if I could leave my templates (that rarely change) in the sip.conf file and put the bare necessities in realtime (users that change all the time). Thanks, Charles Wadsworth ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf wont load completely
14 apr 2008 kl. 16.19 skrev Al lists: I have seen this issue on both 1.2 and 1.4, was not able to reproduce to find a cause or bug. I have seen this after power failure boot up. show sip peer command shows most of peers, except one or two (in my cases trunk) . if i issue a sip reload command, it will show all of them. I can write a script to reload asterisk after a minute of boot up but i wanted to see if anyone else has seen this issue or has any thoughts. Could be a DNS issue. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf wont load completely
I have seen this issue on both 1.2 and 1.4, was not able to reproduce to find a cause or bug. I have seen this after power failure boot up. show sip peer command shows most of peers, except one or two (in my cases trunk) . if i issue a sip reload command, it will show all of them. I can write a script to reload asterisk after a minute of boot up but i wanted to see if anyone else has seen this issue or has any thoughts. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf setvar option
Hi, does anybody know about the setvar option in asterisk's sip.conf. I am trying to define it for a peer that's used when making calls using the originate ami call, but it seems to not have any effect. Marcus -- Marcus Hunger - [EMAIL PROTECTED] Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 indigo networks GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5713/2881, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf setvar option
On Fri, 2008-03-28 at 12:30 +0100, Marcus Hunger wrote:
does anybody know about the setvar option in asterisk's sip.conf.
Sure! This is one of my favorite features.
Let's say I have a definition for my phone in sip.conf, and it looks
something like this:
[myphone]
secret=verysecretpassword
type=friend ; a friend is both a user and a peer
host=dynamic ; phone will register to Asterisk
disallow=all
allow=gsm ; first, try to negotiate gsm
allow=ulaw; the try ulaw
setvar=MYVAR=blah
Whenever a call comes into Asterisk from this particular phone, Asterisk
will automatically create a channel variable named MYVAR, and ${MYVAR}
will contain the value blah. I can then use it for whatever purpose I
see fit within my dialplan.
--
Jared Smith
Community Relations Manager
Digium, Inc.
___
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Re: [asterisk-users] sip.conf setvar option
28 mar 2008 kl. 13.42 skrev Jared Smith:
On Fri, 2008-03-28 at 12:30 +0100, Marcus Hunger wrote:
does anybody know about the setvar option in asterisk's sip.conf.
Sure! This is one of my favorite features.
Let's say I have a definition for my phone in sip.conf, and it looks
something like this:
[myphone]
secret=verysecretpassword
type=friend ; a friend is both a user and a peer
host=dynamic ; phone will register to Asterisk
disallow=all
allow=gsm ; first, try to negotiate gsm
allow=ulaw; the try ulaw
setvar=MYVAR=blah
Whenever a call comes into Asterisk from this particular phone,
Asterisk
will automatically create a channel variable named MYVAR, and ${MYVAR}
will contain the value blah. I can then use it for whatever
purpose I
see fit within my dialplan.
Well, Jared, but that's the reverse. You stripped out this important
part:
am trying to define it for a peer that's used when making calls
using the originate ami call, but it seems to not have any effect.
The important thing with your lesson was that SETVAR is only used on
INCOMING calls from
devices, not outbound calls TO devices. Using ORIGINATE to call a SIP
peer, there's no variables
set from sip.conf.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/ * SIP Masterclass
Orlando FL * April 21-25 2008
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Re: [asterisk-users] sip.conf setvar option
So, wouldn't it be great to enable setvar for outgoing calls too?
On Fri, Mar 28, 2008 at 1:55 PM, Johansson Olle E [EMAIL PROTECTED] wrote:
28 mar 2008 kl. 13.42 skrev Jared Smith:
On Fri, 2008-03-28 at 12:30 +0100, Marcus Hunger wrote:
does anybody know about the setvar option in asterisk's sip.conf.
Sure! This is one of my favorite features.
Let's say I have a definition for my phone in sip.conf, and it looks
something like this:
[myphone]
secret=verysecretpassword
type=friend ; a friend is both a user and a peer
host=dynamic ; phone will register to Asterisk
disallow=all
allow=gsm ; first, try to negotiate gsm
allow=ulaw; the try ulaw
setvar=MYVAR=blah
Whenever a call comes into Asterisk from this particular phone,
Asterisk
will automatically create a channel variable named MYVAR, and ${MYVAR}
will contain the value blah. I can then use it for whatever
purpose I
see fit within my dialplan.
Well, Jared, but that's the reverse. You stripped out this important
part:
am trying to define it for a peer that's used when making calls
using the originate ami call, but it seems to not have any effect.
The important thing with your lesson was that SETVAR is only used on
INCOMING calls from
devices, not outbound calls TO devices. Using ORIGINATE to call a SIP
peer, there's no variables
set from sip.conf.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/ * SIP Masterclass
Orlando FL * April 21-25 2008
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--
Marcus Hunger - [EMAIL PROTECTED]
Telefon: +49 (0)211-63 55 55-61
Telefax: +49 (0)211-63 55 55-22
indigo networks GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5713/2881, Umsatzsteuer-ID: DE219349391
www.sipgate.de - www.sipgate.at - www.sipgate.co.uk
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Re: [asterisk-users] sip.conf setvar option
On Fri, 2008-03-28 at 13:55 +0100, Johansson Olle E wrote: Well, Jared, but that's the reverse. You stripped out this important part: am trying to define it for a peer that's used when making calls using the originate ami call, but it seems to not have any effect. The important thing with your lesson was that SETVAR is only used on INCOMING calls from devices, not outbound calls TO devices. Using ORIGINATE to call a SIP peer, there's no variables set from sip.conf. Absolutely true... and I'll make up for it by pointing out that if you're using the Originate manager command, you can set channel variables by adding the Variable setting to your manager command: Action: Originate Channel: SIP/myphone Context: test Exten: 123 Priority: 1 Async: True ActionID: ThisIsMyVeryOriginalActionID Variable: MYVAR=blah|ANOTHERVAR=baz -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf setvar option
28 mar 2008 kl. 14.00 skrev Marcus Hunger: So, wouldn't it be great to enable setvar for outgoing calls too? Well, maybe in the outbound channel then. But that won't help much. mixing the caller's and callee's variables in the INCOMING channel would be messy and only cause issues. But there's another way. Hint hint. Friday afternoon hack. /O ;-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf setvar option
Particularly, I want to set the SIPADDHEADER variable dynamicly for peers with rt-engine. Working around it might be possible, but having the thing working transparently for Dial and Originate would be great. On Fri, Mar 28, 2008 at 2:47 PM, Johansson Olle E [EMAIL PROTECTED] wrote: 28 mar 2008 kl. 14.00 skrev Marcus Hunger: So, wouldn't it be great to enable setvar for outgoing calls too? Well, maybe in the outbound channel then. But that won't help much. mixing the caller's and callee's variables in the INCOMING channel would be messy and only cause issues. But there's another way. Hint hint. Friday afternoon hack. /O ;-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Marcus Hunger - [EMAIL PROTECTED] Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 indigo networks GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5713/2881, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf setvar option
28 mar 2008 kl. 14.56 skrev Marcus Hunger: Particularly, I want to set the SIPADDHEADER variable dynamicly for peers with rt-engine. Working around it might be possible, but having the thing working transparently for Dial and Originate would be great. That should work today with the unofficial backdoor I implemented. sipaddheader just adds a few channel variables that the outbound channel inherits. If you add them yourself with setvar=_SIPADDHEADER99=X-peeraccountcode: 12345 I think that should work. Out of the box, like magic. This of course only works with calls FROM peers. Have a nice weekend! /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf setvar option
Ok,
Now I have a friday afternoon patch for the community.
In the branch
http://svn.digium.com/view/asterisk/team/oej/peer-chanvars/
there's an addition to the SIPPEER() dialplan function where you can
retrieve a setvar= channel variable defined in sip.conf for the peer.
The branch is based on 1.4 and the patch will soon be included in the
1.6 trunk.
This way, you can for example add a variable called CELLPHONE with
the peer's cell phone number. If dial(sip/olle) fails, I can now do
dial(zap/${SIPPEER(olle,chanvar[CELLPHONE])})
This is something I came up with a few weeks ago when I created a PBX
based on Asterisk for a company, something that I don't do much, since
I normally use Asterisk in carrier environments with SIP proxys. Doing
this little PBX project was a lot of fun and I learned a lot.
Have a nice weekend!
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Asterisk SIP Masterclass, Orlando Florida April 21-25 2008. Register
today!
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[asterisk-users] sip.conf help, inbound calls fall to last specified context
First of all, if Asterisk is the client and it must register to the other side, does the peer\user entry have to be in sip.conf, or can it be in ARA? Second, why do all calls fall through to the last context specified, whether in that peer\user definition or not? I'm assuming it's a typo somewhere, but I can't find it. I had a full sip.conf, but axed a lot of the fluff trying to remove any source of typo. [EMAIL PROTECTED] asterisk]# cat sip.conf [general] context=default port=5060 canreinvite=no ;register = 8157582715::[EMAIL PROTECTED] ; ottos 815-758-2715 register = 8157879826::[EMAIL PROTECTED] ; ottos 815-787-9826 ;register = 8159092441::[EMAIL PROTECTED] ; RWest 815-909-2441 ;register = 8159092443::[EMAIL PROTECTED] ; RWest 815-909-2443 ;- REALTIME SUPPORT ; For additional information on ARA, the Asterisk Realtime Architecture, ; please read realtime.txt and extconfig.txt in the /doc directory of the ; source code. ; rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis? (yes|no) ;rtsavesysname=yes ; Save systemname in realtime database at registration ; Default= no ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) ; If set to yes, when a SIP UA registers successfully, the ip address, ; the origination port, the registration period, and the username of ; the UA will be set to database via realtime. ; If not present, defaults to 'yes'. ;rtautoclear=yes; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered? (yes|no|seconds) ; If set to yes, when the registration expires, the friend will ; vanish from the configuration until requested again. If set ; to an integer, friends expire within this number of seconds ; instead of the registration interval. ;ignoreregexpire=yes; Enabling this setting has two functions: ; ; For non-realtime peers, when their registration expires, the ; information will _not_ be removed from memory or the Asterisk database ; if you attempt to place a call to the peer, the existing information ; will be used in spite of it having expired ; ; For realtime peers, when the peer is retrieved from realtime storage, ; the registration information will be used regardless of whether ; it has expired or not; if it expires while the realtime peer ; is still in memory (due to caching or other reasons), the ; information will not be removed from realtime storage [8157582715] type=friend accountcode=2 context=ottos secret= username=2715 fromuser=8157582715 insecure=very host=63.175.151.3 ;voip.essex1.com fromdomain=63.175.151.3 ;voip.essex1.com canreinvite=no disallow=all allow=ulaw allow=alaw ;qualify=yes [8159092441] type=friend accountcode=12 context=rwest secret= username=2441 fromuser=8159092441 insecure=very host=63.175.151.3 ;voip.essex1.com fromdomain=63.175.151.3 ;voip.essex1.com canreinvite=no disallow=all allow=ulaw allow=alaw ;qualify=yes [8159092443] type=friend accountcode=12 context=rwest secret= username=2441 fromuser=8159092443 insecure=very host=63.175.151.3 ;voip.essex1.com fromdomain=63.175.151.3 ;voip.essex1.com canreinvite=no disallow=all allow=ulaw allow=alaw ;qualify=yes [8157879826] type=friend ;accountcode=2 context=ics secret= username=9826 fromuser=8157879826 insecure=very host=63.175.151.3 ;voip.essex1.com fromdomain=63.175.151.3 ;voip.essex1.com ;canreinvite=no ;disallow=all ;allow=ulaw -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf help, inbound calls fall to last specified context
Updated with a smaller sip.conf that also doesn't work right. [EMAIL PROTECTED] asterisk]# cat sip.conf [general] port=5060 canreinvite=no rtcachefriends=yes disallow=all allow=ulaw allow=alaw register = 8157879826::[EMAIL PROTECTED] ; ottos 815-787-9826 register = 8159092443::[EMAIL PROTECTED] ; RWest 815-909-2443 [8157879826] type=friend accountcode=2 context=ics secret= username=9826 fromuser=8157589826 insecure=very host=voip.essex1.com fromdomain=voip.essex1.com [8159092443] type=friend accountcode=12 context=rwest secret= username=2441 fromuser=8159092443 insecure=very host=63.175.151.3 ;voip.essex1.com fromdomain=63.175.151.3 ;voip.essex1.com -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett To: asterisk-users@lists.digium.com Sent: Thursday, March 13, 2008 9:13 AM Subject: [asterisk-users] sip.conf help,inbound calls fall to last specified context First of all, if Asterisk is the client and it must register to the other side, does the peer\user entry have to be in sip.conf, or can it be in ARA? Second, why do all calls fall through to the last context specified, whether in that peer\user definition or not? I'm assuming it's a typo somewhere, but I can't find it. I had a full sip.conf, but axed a lot of the fluff trying to remove any source of typo. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf realtime
At the moment in order to register you must use static configs in sip.conf. As far as the friend/peer etc. settings you can do that in sip.conf or in real time. You can also mix and match (use real time and static configuration). - Original Message - From: hugolivude To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, December 29, 2007 12:10 AM Subject: [asterisk-users] sip.conf realtime Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use realtime static for the [general] part of my sip.conf including the registration commands: register=did:secret@domain/did context and use realtime realtime (funny name!) for peers and friends: [myprovider] type=peer auth=md5 username=... fromuser=... fromdomain=... secret=... host=... port=5060 nat=yes canreinvite=yes qualify=no disallow=all allow=ulaw dtmfmode=rfc2833 insecure=port,invite context=incoming-sip Is this correct? What's throwing me off is this statment found here: NOTE: You can only store a static config OR a RealTime config. You cannot, for example, store sip.conf and use sipfriends via RealTime. This would suggest that I'll have to do a reload when I add a DiD, but a reload won't be necessary if a new SIP client is added. Do I have it right? Also, what's the difference between a peer and a user? I used to think that a user was an agent authorized to call in to my * box, a peer was an agent I could reach and a freind was both. What's throwing me off now is the statement found here: With newer versions of Asterisk the concept of SIP 'users' will be phased out. I can't understand this especially in the context of extconfig.conf that uses both a sipuser and sippeer entry. Could someone clarify for me? Thanks, H -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf for internetcalls.com
404 means that the number you are dialing is not available on the remote end. Is there anything that you do that makes it break or is it random ? If it is random I would speak to your ITSP. - Original Message - From: Jaap Winius [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, December 25, 2007 1:14 AM Subject: Re: [asterisk-users] sip.conf for internetcalls.com Quoting Justin Case [EMAIL PROTECTED]: What comes up in the Asterisk CLI? When it's not working, nothing appears in the CLI even though I've used set verbose 10. Also it can be a NAT issue? How can that lead to this intermittent behavior? I've already set nat=yes. Also, I'm using an ADSL router with a NAT; not anything like iptables. Have Asterisk register every 3-4 minutes. I'm not sure how to do that. I found defaultexpirey, but the default for it is two minutes. Anyway, why would that help with Asterisk, when my previous SIP client, a Linksys SPA3000, was configured with a register expire time of an hour and worked fine with InternetCalls.com. I think something else is going on. Using tcpdump, I see this when things are working okay: -- 23:38:05.354523 IP 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip bitis.umrk.to.sip: SIP, length: 847 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via 23:38:05.355065 IP bitis.umrk.to.sip 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip: SIP, length: 471 FSIP/2.0 100 Trying Via: SIP/2.0/UDP 194.221.62.198:50 . 23:38:11.007350 IP 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip bitis.umrk.to.sip: SIP, length: 507 ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: S -- The ACK packet is sent after the conversation (.) has ended. However, when it doesn't work, I see this: -- 23:42:24.736377 IP 194.120.0.198.sip bitis.umrk.to.sip: SIP, length: 841 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via 23:42:24.736898 IP bitis.umrk.to.sip 194.120.0.198.sip: SIP, length: 445 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 194.120.0.198: 23:42:24.756967 IP 194.120.0.198.sip bitis.umrk.to.sip: SIP, length: 505 ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: S -- In this case, the ACK follows immediately after the 404 Not Found. Cheers, Jaap ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf realtime
Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use realtime static for the [general] part of my sip.conf including the registration commands: register=did:secret@domain/did context and use realtime realtime (funny name!) for peers and friends: [myprovider] type=peer auth=md5 username=... fromuser=... fromdomain=... secret=... host=... port=5060 nat=yes canreinvite=yes qualify=no disallow=all allow=ulaw dtmfmode=rfc2833 insecure=port,invite context=incoming-sip Is this correct? What's throwing me off is this statment found here:http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static * NOTE:* You can only store a static config OR a RealTime config. You cannot, for example, store sip.conf and use sipfriends via RealTime. This would suggest that I'll have to do a reload when I add a DiD, but a reload won't be necessary if a new SIP client is added. Do I have it right? Also, what's the difference between a peer and a user? I used to think that a user was an agent authorized to call in to my * box, a peer was an agent I could reach and a freind was both. What's throwing me off now is the statement found here:http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static With newer versions of Asterisk the concept of SIP 'users' will be phased out. I can't understand this especially in the context of extconfig.conf that uses both a sipuser and sippeer entry. Could someone clarify for me? Thanks, H ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf for internetcalls.com
Hi all, Perhaps someone here could help me with this. I'm new to Asterisk, but have already met with some success at getting my first system to work with two different VoIP (SIP) providers: XS4ALL and InternetCalls.com. The config for the former works fine, but my InternetCalls.com config works only intermittently for incoming calls. It currently looks like this: [general] port=5060 bindaddr=0.0.0.0 srvlookup=yes register = usr-name:passwd@sip.internetcalls.com/inetcalls-in [inetcalls] type=friend context=inetcalls-in nat=yes username=usr-name fromuser=usr-name secret=passwd host=sip.internetcalls.com canreinvite=no qualify=yes dtmfmode=inband insecure=invite disallow=all allow=ulaw allow=alaw When I dial in from outside and it doesn't work, all I get is a busy signal and no further clues as to what might be wrong. Any ideas? Thanks very much, Jaap ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf for internetcalls.com
What comes up in the Asterisk CLI ? Set debug and verbosity to 9 and see what comes up. Also it can be a NAT issue ? Have Asterisk register every 3-4 minutes. On Dec 24, 2007 4:00 PM, Jaap Winius [EMAIL PROTECTED] wrote: Hi all, Perhaps someone here could help me with this. I'm new to Asterisk, but have already met with some success at getting my first system to work with two different VoIP (SIP) providers: XS4ALL and InternetCalls.com. The config for the former works fine, but my InternetCalls.com config works only intermittently for incoming calls. It currently looks like this: [general] port=5060 bindaddr=0.0.0.0 srvlookup=yes register = usr-name:passwd@sip.internetcalls.com/inetcalls-in [inetcalls] type=friend context=inetcalls-in nat=yes username=usr-name fromuser=usr-name secret=passwd host=sip.internetcalls.com canreinvite=no qualify=yes dtmfmode=inband insecure=invite disallow=all allow=ulaw allow=alaw When I dial in from outside and it doesn't work, all I get is a busy signal and no further clues as to what might be wrong. Any ideas? Thanks very much, Jaap ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf for internetcalls.com
Quoting Justin Case [EMAIL PROTECTED]: What comes up in the Asterisk CLI? When it's not working, nothing appears in the CLI even though I've used set verbose 10. Also it can be a NAT issue? How can that lead to this intermittent behavior? I've already set nat=yes. Also, I'm using an ADSL router with a NAT; not anything like iptables. Have Asterisk register every 3-4 minutes. I'm not sure how to do that. I found defaultexpirey, but the default for it is two minutes. Anyway, why would that help with Asterisk, when my previous SIP client, a Linksys SPA3000, was configured with a register expire time of an hour and worked fine with InternetCalls.com. I think something else is going on. Using tcpdump, I see this when things are working okay: -- 23:38:05.354523 IP 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip bitis.umrk.to.sip: SIP, length: 847 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via 23:38:05.355065 IP bitis.umrk.to.sip 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip: SIP, length: 471 FSIP/2.0 100 Trying Via: SIP/2.0/UDP 194.221.62.198:50 . 23:38:11.007350 IP 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip bitis.umrk.to.sip: SIP, length: 507 ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: S -- The ACK packet is sent after the conversation (.) has ended. However, when it doesn't work, I see this: -- 23:42:24.736377 IP 194.120.0.198.sip bitis.umrk.to.sip: SIP, length: 841 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via 23:42:24.736898 IP bitis.umrk.to.sip 194.120.0.198.sip: SIP, length: 445 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 194.120.0.198: 23:42:24.756967 IP 194.120.0.198.sip bitis.umrk.to.sip: SIP, length: 505 ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: S -- In this case, the ACK follows immediately after the 404 Not Found. Cheers, Jaap ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf best practices?
We use the MAC of the phone (all lower case) with a -a, -b, -c, etc tacked onto the end of the MAC to specify the line appearance. One thing you MUST remember is that a sip.conf entry is NOT an extension. Extensions are totally different from sip.conf entries. sip.conf entries are DEVICES. Using the extension as the SIP account ID will cause MORE work as you have requests for oddball and custom call routing. We do something like this as an extension: exten = 3727,1,Set([EMAIL PROTECTED]) exten = 3727,n,Set(DIAL_DEST=SIP/0004f211f9a6-a) exten = 3727,n,Set(CFBL_DEST=SIP/0004f211f9a6-b) exten = 3727,n,Macro(std-exten-v2) The std-exten-v2 handles the call based on the variables set before the macro is run. Here is another example: exten = 3733,1,Set(DIAL_DEST=SIP/0004f2127e9c-a) exten = 3733,n,Set(CFBL_DEST==SIP/0004f2127e9c-b) exten = 3733,n,Set(CFNA_DEST=Local/[EMAIL PROTECTED]) exten = 3733,n,Set(OPER_DEST=Local/[EMAIL PROTECTED]) exten = 3733,n,Set([EMAIL PROTECTED]) exten = 3733,n,Macro(std-exten-v2) Telecom newbies seem to think dialplans can be simple or short or elegant. They are not. Dialplans are ugly and complex beasts. Users want custom call routing, users want this, users want that, etc. No matter how hard you try to make your dialplan simple, it won't stay that way. Erik Anderson wrote: All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and routing calls through this server. The question is what to use for the username portion of the SIP account. Part of me says that I should standardize on using each phone's MAC address as the sip account UID, like so: ; Joe Smith, x123 [000E08DA0409] secret = blahblah ... and so on and so forth Doing it that way is nice for standardization's sake, but it makes the dialplan quite a bit more complex. The obvious alternative is to use the extension as the sip UID: ; Joe Smith, x123 [123] secret = blahblah ... This makes the dialplan *much* more simple, but when looking through sip.conf, it's not as immediately obvious what device should be authenticating with that account. Since this is my first large-ish asterisk deployment, I'm seeking the advice of those who have gone before me. What tactic (one of the above options or otherwise) is best to keep your sip.conf sane? Thanks! -Erik ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf best practices?
Erik Anderson wrote: All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and routing calls through this server. The question is what to use for the username portion of the SIP account. Part of me says that I should standardize on using each phone's MAC address as the sip account UID, like so: ; Joe Smith, x123 [000E08DA0409] secret = blahblah ... and so on and so forth Doing it that way is nice for standardization's sake, but it makes the dialplan quite a bit more complex. The obvious alternative is to use the extension as the sip UID: ; Joe Smith, x123 [123] secret = blahblah ... This makes the dialplan *much* more simple, but when looking through sip.conf, it's not as immediately obvious what device should be authenticating with that account. Since this is my first large-ish asterisk deployment, I'm seeking the advice of those who have gone before me. What tactic (one of the above options or otherwise) is best to keep your sip.conf sane? Thanks! -Erik Hi Erik, We have around 100 devices and most of our changes are adds for new users/devices with occasional re-assignment of devices. We manage our users and devices with some simple scripts and good old vi for exceptions. Our extensions.conf has a list of global vars that tie an extension to a sip (or iax, or whatever) device. (I think this is straight from TFOT v1) eg EXT_100=SIP/100 This allows us to redirect extensions to different devices or make extensions ring on multiple devices by changing that var alone (no need to alter macros or other dialplan elements) eg EXT_100=SIP/101SIP/102 The device-specific hardware and the SIP configurations are generated from a master map that contains a line per device including technology and extension, MAC, user display name and email address. Scripts create the phone hardware configs with device type determined by MAC address (eg. Aastra, Grandstream or Cisco) from the map file and add the user to sip.conf and voicemail.conf. sip.conf ... [grandstream] ; Aastra 480i phones for general office ... (general SIP settings) context=office-dial [aastra-cc] ; Aastra 480i phones for Call Centre only ... (general SIP settings) context=cc-dial [100](grandstream) username=100 secret=*** mailbox=100 callerid=Joe Bloggs 100 [101](aastra-cc) username=101 secret=*** mailbox=101 callerid=Agent 99 101 Initially we only had one class of user, general office types using Grandstreams. When we migrated the Call Centre to Asterisk, they were the only users with Aastra phones, apart from one or two in the general office. So each class of user had a different hardware type and was easy to automate, the exceptions are currently handled with manual edits. This system is working well and is stable but not sufficiently flexible for the future. Our company is growing rapidly and since we will no longer be buying Grandstream devices, more Aastras are appearing in the general office environment. This means we now have two classes of users that require different configs for the same device type, general office users and Call Centre agents. This means a choice will have to be made between updating the scripts to cope with the two user classes or moving to realtime. Where's my Magic 8-ball... :-) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf best practices?
All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and routing calls through this server. The question is what to use for the username portion of the SIP account. Part of me says that I should standardize on using each phone's MAC address as the sip account UID, like so: ; Joe Smith, x123 [000E08DA0409] secret = blahblah ... and so on and so forth Doing it that way is nice for standardization's sake, but it makes the dialplan quite a bit more complex. The obvious alternative is to use the extension as the sip UID: ; Joe Smith, x123 [123] secret = blahblah ... This makes the dialplan *much* more simple, but when looking through sip.conf, it's not as immediately obvious what device should be authenticating with that account. Since this is my first large-ish asterisk deployment, I'm seeking the advice of those who have gone before me. What tactic (one of the above options or otherwise) is best to keep your sip.conf sane? Thanks! -Erik -- Erik Anderson http://andersonfam.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf best practices?
Use the extension, and use grep to determine which account uses which phone. For example I provision my spa9xx phones from a subdirectory on apache called spa which on slackware is at: /var/www/htdocs/spa/ doing: grep 123 /var/www/htdocs/spa/* will tell you which phone it is. On 9/18/07, Erik Anderson [EMAIL PROTECTED] wrote: All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and routing calls through this server. The question is what to use for the username portion of the SIP account. Part of me says that I should standardize on using each phone's MAC address as the sip account UID, like so: ; Joe Smith, x123 [000E08DA0409] secret = blahblah ... and so on and so forth Doing it that way is nice for standardization's sake, but it makes the dialplan quite a bit more complex. The obvious alternative is to use the extension as the sip UID: ; Joe Smith, x123 [123] secret = blahblah ... This makes the dialplan *much* more simple, but when looking through sip.conf, it's not as immediately obvious what device should be authenticating with that account. Since this is my first large-ish asterisk deployment, I'm seeking the advice of those who have gone before me. What tactic (one of the above options or otherwise) is best to keep your sip.conf sane? Thanks! -Erik -- Erik Anderson http://andersonfam.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf best practices?
Realtime and sip_buddies in mysql works well for very large installations. PaulH On Tue, 2007-09-18 at 22:11 -0500, Erik Anderson wrote: All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and routing calls through this server. The question is what to use for the username portion of the SIP account. Part of me says that I should standardize on using each phone's MAC address as the sip account UID, like so: ; Joe Smith, x123 [000E08DA0409] secret = blahblah ... and so on and so forth Doing it that way is nice for standardization's sake, but it makes the dialplan quite a bit more complex. The obvious alternative is to use the extension as the sip UID: ; Joe Smith, x123 [123] secret = blahblah ... This makes the dialplan *much* more simple, but when looking through sip.conf, it's not as immediately obvious what device should be authenticating with that account. Since this is my first large-ish asterisk deployment, I'm seeking the advice of those who have gone before me. What tactic (one of the above options or otherwise) is best to keep your sip.conf sane? Thanks! -Erik ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf best practices?
On 9/18/07, C F [EMAIL PROTECTED] wrote: Use the extension, and use grep to determine which account uses which phone. For example I provision my spa9xx phones from a subdirectory on apache called spa which on slackware is at: /var/www/htdocs/spa/ doing: grep 123 /var/www/htdocs/spa/* will tell you which phone it is. That's a great idea - probably seems like the most simple option. Thanks! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf best practices?
The obvious alternative is to use the extension as the sip UID: Use the extension as the UID and add the mac address as a comment. Like so: [123] ; Joe Smith ;mac=000E08DA0409 secret = blahblah ... and so on and so forth This will give the best of both worlds. The mac is readily available and the dialplan is clear. I usually try to go one further and setup dhcp to set the last octet of the IP address to the extension number. This makes it easy to point a browser to the phone for configuration as well. John ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP.CONF: incominglimit and outgoinglimit
Hi all, I have some peers configured in SIP.CONF file with parameters incominglimit and outgoinglimit set up to 10. By doing that, I expect that this peer will not be allowed to handle more than 10 incoming calls and 10 outgoing calls at the same time. However, since I upgraded to Asterisk 1.2.17, I started to face a problem. Sometimes, calls to those peers are not connected. When I check the logs, the following message is displayed: 2007-05-23 10:25:49 NOTICE[9630]: chan_sip.c:2273 update_call_counter: Call to peer '00.peer101' rejected due to usage limit of 10 Checking the open channels (CLIshow channels), I can see that there are no calls assigned to this peer, so this message makes no sense. Also, the only way to fix it is increase the parameters incominglimit and outgoinglimit and reload OR restart Asterisk. I read something about End of Life for [outgoinglimit and incominglimit] announced, please use setgroup and checkgroup Questions is: should I still use outgoinglimit and incominglimit in Asterisk 1.2.17? If so, what is the correct CLI command to confirm that we reached the call limit for a given peer? If not, what parameters should I use now? Thanks. Fernando ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf limitonpeers=yes in asterisk 1.4
Hi, An observation on this feature, which I may have completely misunderstood, so flame away if I am being dumb :) Looking at the code, setting limitonpeers=yes causes all user and peer calls to be ref-counted as if they are peer calls (assuming a user and peer of the same name exist). A side-effect of this is that an incoming call seems to have its call-limit evaluated based on the peer's, rather than the user's settimg, unless no call-limit has been set against the user, in which case the peer's call-limit is ignored too. I also noticed that if an inbound (user) call is blocked based on the above, then unref_peer(p) is called, instead of unref_user(u) - I have no clue what that does, so it may be quite safe :) Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf - srvlookup
I would like to put srvlookup=no in my SIP conf, so that I don't get DNS issues (Asterisk stops responding). I use VoIP Buster and in sip.conf I use sip1.voipbuster.com. When I do sip show peers in CLI I get voipbuster/tomo 194.221.62.207 5060 OK (27 ms) And when I ping sip1.voipbuster.com [EMAIL PROTECTED] ~]# ping sip1.voipbuster.com PING sip1.voipbuster.com (194.221.62.206) 56(84) bytes of data. So, Asterisk is registered at 194.221.62.207 and DNS lookup gives me 194.221.62.206 IP address. Question is, which IP address should I use, 206 or 207? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf for talking to other Asterisk machines
Just curious how most of you are defining SIP peers in sip.conf for Asterisk boxes talking to each other. Are most of you just making a type=friend connection and a single context or are you separating them out to in/out definitions and contexts? In other words Where voicegw1 is the Asterisk box with the TDM cards for talking to the PSTN, it will receive calls from the PSTN and forward to the appropriate Asterisk box as well as receive calls from the other Asterisk boxes to forward out to the PSTN. Do you on the Asterisk box that contains all the SIP phones define (ie the client to the PSTN Asterisk box and voicegw1 is the one with the PSTN connection) [voicegw1-in] type=user username=virtualpbx1-in secret=1234 host=192.168.1.99 context=voicegw1-in canreinvite=no nat=no qualify=yes allow=all [voicegw1-out] type=peer username=virtualpbx1-out secret=1234 host=192.168.1.99 context=voicegw1-out canreinvite=no nat=no qualify=yes allow=all or [voicegw1] Type=friend Blah Context=voicegw1 And use a single context for inbound/outbound routing? The same would apply to the PSTN Asterisk server. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf for talking to other Asterisk machines
I use two user's per host one for user and the other peer. Sort of like attahed. I also prefer IAX for communication between asterisk boxes. IAX use's less bandwidth than SIP and it's trunks are alot smaller. If you look at SIP traffic, 80% of it is headers. The headers look just like smtp headers. Even if your clients are using SIP to communicate to asterisk using SIP, the asterisk servers will maintain the trunked connection route the traffic for your SIP phones. On 9/18/06, Bill Gibbs [EMAIL PROTECTED] wrote: Just curious how most of you are defining SIP peers in sip.conf – for Asterisk boxes talking to each other. Are most of you just making a type=friend connection and a single context or are you separating them out to in/out definitions and contexts? In other words Where voicegw1 is the Asterisk box with the TDM cards for talking to the PSTN, it will receive calls from the PSTN and forward to the appropriate Asterisk box as well as receive calls from the other Asterisk boxes to forward out to the PSTN. Do you on the Asterisk box that contains all the SIP phones define (ie the client to the PSTN Asterisk box and voicegw1 is the one with the PSTN connection) [voicegw1-in] type=user username=virtualpbx1-in secret=1234 host=192.168.1.99 context=voicegw1-in canreinvite=no nat=no qualify=yes allow=all [voicegw1-out] type=peer username=virtualpbx1-out secret=1234 host=192.168.1.99 context=voicegw1-out canreinvite=no nat=no qualify=yes allow=all or [voicegw1] Type=friend Blah Context=voicegw1 And use a single context for inbound/outbound routing? The same would apply to the PSTN Asterisk server. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 662006_23640_0.png Description: PNG image ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] sip.conf for talking to other Asterisk machines
IAX has some pretty severe limitations when it comes to trunking calls between Asterisk boxes. It can't pass variables for example, and any calls to SIP phones at the far end will be treated as IAX calls, which is just nuts. This means you lose a lot of SIP features, like transferring and forwarding. We had to drop IAX and go back to SIP, which is pretty ironic considering IAX stands for Inter Asterisk Exchange. -Original Message- From: Forrest Beck [mailto:[EMAIL PROTECTED] Sent: Mon 9/18/2006 7:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] sip.conf for talking to other Asterisk machines I use two user's per host one for user and the other peer. Sort of like attahed. I also prefer IAX for communication between asterisk boxes. IAX use's less bandwidth than SIP and it's trunks are alot smaller. If you look at SIP traffic, 80% of it is headers. The headers look just like smtp headers. Even if your clients are using SIP to communicate to asterisk using SIP, the asterisk servers will maintain the trunked connection route the traffic for your SIP phones. On 9/18/06, Bill Gibbs [EMAIL PROTECTED] wrote: Just curious how most of you are defining SIP peers in sip.conf – for Asterisk boxes talking to each other. Are most of you just making a type=friend connection and a single context or are you separating them out to in/out definitions and contexts? In other words Where voicegw1 is the Asterisk box with the TDM cards for talking to the PSTN, it will receive calls from the PSTN and forward to the appropriate Asterisk box as well as receive calls from the other Asterisk boxes to forward out to the PSTN. Do you on the Asterisk box that contains all the SIP phones define (ie the client to the PSTN Asterisk box and voicegw1 is the one with the PSTN connection) [voicegw1-in] type=user username=virtualpbx1-in secret=1234 host=192.168.1.99 context=voicegw1-in canreinvite=no nat=no qualify=yes allow=all [voicegw1-out] type=peer username=virtualpbx1-out secret=1234 host=192.168.1.99 context=voicegw1-out canreinvite=no nat=no qualify=yes allow=all or [voicegw1] Type=friend Blah Context=voicegw1 And use a single context for inbound/outbound routing? The same would apply to the PSTN Asterisk server. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf, extensions.conf
Hi to all, I am just trying for Asterisk on my fedora core3 machine. here's my sip extensions config files sip.conf [general] bindport=5060 bindaddr=0.0.0.0 allow=all context=ECPT localnet=192.168.0.1 localmask=255.255.255.0 [phone1] type=friend host=192.168.0.53 context=Embedded callerid=ashok449 [phone2] type=friend host=192.168.0.22 context=Embedded callerid=pramod450 [phone3] type=friend host=192.168.0.54 context=Embedded callerid=pramod451 extensions.conf [general] static=yes writeprotect=yes [globals] CONSOLE=Console/dsp [local] include=default include=Embedded include=ECPT [ECPT] exten=-.,1,congestion [Embedded] exten=449,1,dial(SIP/phone1,20) exten=449,2,voicemail,u449 exten=450,1,dial(SIP/phone2,20) exten=450,2,voicemail,u450 exten =451,1,dial(SIP/phone3,20) exten =451,2,voicemail,u451 I need you people help for following things 1) how do i configure sip.conf to register my clients when host=dynamic? 2) how do i configure extensions.conf to make a call to PSTN ? 3) suggest a Softphone for asterisk on LAN. 4) how to establish a connection between two Asterisk servers? thanks in advance, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip.conf: domain=huh?
So I saw mention of a way to allow dialing using SIP URI's on Dave McNett's site at http://slacker.com/~nugget/projects/asterisk/page7 Wow, awesome, I can call anywhere now. However, I think there is a more elegant way of figuring out whether or not the local * server should handle a given domain. Specifically, Dave compares a series of domains within extensions.conf to figure out how to handle the call. I would rather use something like the function CHECKSIPDOMAIN - this feels more appropriate. So, CHECKSIPDOMAIN uses the domain=blah entries in sip.conf. However, contrary to the docs, it would seem that one MUST associate a context with each domain= entry. My * complains otherwise with a warning upon load that the domains are not associated with any contexts, and when I run sip show domains at the CLI I get SIP Domain support not enabled. I am hesitant to set that up, too. Does anyone understand the security implications of domain= entries in sip.conf? Especially since I want to use these entries only for deciding how to handle outbound calls. I guess what I am saying is that since this is for outbound calls, and I need to specify a context, how does this affect incoming calls and the specified context from a security point of view? What will domain= change when trying to authenticate calls? Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip.conf codecs: ulaw, alaw and g729
Hi, When ever I put g729 in allow for trunk the other two codecs (ulaw and alaw) stop working and I get the frame type error for them, but g729 works fine. I've cleared general part of sip.conf of codec info to be on safe side. If ulaw and alaw are the only ones allowed they work fine. Asterisk shouldn't be doing any encoding or decoding, all codecs should be passing through. Any ideas how I can get all three codecs working in sip.conf/asterisk? example of frame error I get if I use different codec to g729: Apr 20 11:57:14 WARNING[13028]: chan_sip.c:2527 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Apr 20 11:57:14 WARNING[13028]: chan_sip.c:2527 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) sip.conf (with the following g729 works but alaw and ulaw don't): [trunk] dtmfmode=info context=from-outside type=friend host=x.x.x.x ;disallow=all ;allow=g729 ;allow=alaw ;allow=ulaw [general] port=5060 bindaddr=0.0.0.0 insecure=very ;disallow=none ;allow=g729 ;allow=alaw ;allow=ulaw ;allow=gsm context=from-outside Thanks -- Shaun *** If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip.conf
In sip.conf, how can I define that only IP phones from 192.168.0.0/24 network can register with specific user? The thing is that I can't use password and I can't use host=ip.of.my.phone. And I have to be sure that no one, from Internet will register on my * like that user. So, please tell me how to do this? -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip.conf
Hi Check this setting: bindaddr = 0.0.0.0 :IP Address to bind to (listen on) kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Par?ina Sent: Tuesday, April 18, 2006 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sip.conf In sip.conf, how can I define that only IP phones from 192.168.0.0/24 network can register with specific user? The thing is that I can't use password and I can't use host=ip.of.my.phone. And I have to be sure that no one, from Internet will register on my * like that user. So, please tell me how to do this? -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip.conf
In sip.conf, how can I define that only IP phones from 192.168.0.0/24 network can register with specific user? The thing is that I can't use password and I can't use host=ip.of.my.phone. And I have to be sure that no one, from Internet will register on my * like that user. So, please tell me how to do this? Try this in sip.conf under a phone definition: deny=0.0.0.0/0.0.0.0 permit=some_ip_address/some_mask Ivan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip.conf
2006/4/18, Tomislav Parčina [EMAIL PROTECTED]: In sip.conf, how can I define that only IP phones from 192.168.0.0/24 network can register with specific user? The thing is that I can't use password and I can't use host=ip.of.my.phone. And I have to be sure that no one, from Internet will register on my * like that user. So, please tell me how to do this? Asterisk can bind only ips from internal but I think the best way is to configure some firewall rules in your linux box. It is convenient to drop or reject all communications except that you want to accept (http, smtp, etc.). -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP.conf Technical Documentation - Help
Is there a document/wiki/web site that maps the various SIP.conf settings to the structure of the actual IP packet? If so please advise. -- ___ Play 100s of games for FREE! http://games.mail.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP.conf Technical Documentation - Help
http://www.voip-info.org/wiki-asterisk http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf On 12/8/05, John Voss [EMAIL PROTECTED] wrote: Is there a document/wiki/web site that maps the various SIP.conf settings to the structure of the actual IP packet? If so please advise. -- ___ Play 100s of games for FREE! http://games.mail.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
