[asterisk-users] set different codec for different sip calls
hello every one i want to have multiple sip calls with different codecs for each one. for example call to 8100 has g729 codec while call to 7900 has ulaw codec. i searched a lot and found that there is some variable like sip_codec which can set codec for a special inbound or outbound call. i don't try it yet because i prefer to set the codec for each call by setting it in contexts in sip.conf or sip_additional.conf file. is it possible?? if yes, how should i set codec for each context? if not, setting the codec in dial-plans in extensions.conf file, is the only way??? thanks in advance SAM -- _ -- 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] No matching peers message has gone (1.8.23.1)
Hi Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer get the 'no matching peer' error when we get a dictionary SIP attack. Now the logs always show a 'wrong password' when there actually isn't a matching peer. We even have alwaysauthreject = yes in our sip.conf. Has anyone else noticed this phenomenon? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] No matching peers message has gone (1.8.23.1)
Ishfaq Malik wrote: Hi Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer get the 'no matching peer' error when we get a dictionary SIP attack. Now the logs always show a 'wrong password' when there actually isn't a matching peer. We even have alwaysauthreject = yes in our sip.conf. Has anyone else noticed this phenomenon? This is on purpose. To fix some exposure issues the code was changed to have an internal peer (albeit one that can never successfully be authenticated against) that gets used if no real peer is found. This reduces the chance (by a lot) of the code exposing information in some off nominal cases. -- Joshua Colp Digium, Inc. | Senior Software Developer 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
Re: [asterisk-users] No matching peers message has gone (1.8.23.1)
Hi Thanks for the quick response. I'll read all the change logs from now on, I promise! Ish On 4 November 2013 15:29, Joshua Colp jc...@digium.com wrote: Ishfaq Malik wrote: Hi Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer get the 'no matching peer' error when we get a dictionary SIP attack. Now the logs always show a 'wrong password' when there actually isn't a matching peer. We even have alwaysauthreject = yes in our sip.conf. Has anyone else noticed this phenomenon? This is on purpose. To fix some exposure issues the code was changed to have an internal peer (albeit one that can never successfully be authenticated against) that gets used if no real peer is found. This reduces the chance (by a lot) of the code exposing information in some off nominal cases. -- Joshua Colp Digium, Inc. | Senior Software Developer 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] No matching peers message has gone (1.8.23.1)
Hi Ish,
I assume you are using Fail2Ban to monitor the logs for dictionary attacks - If
so, the following regex should work for 1.8:
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Wrong password
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - No matching peer
found
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Username/auth name
mismatch
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Device does not
match ACL
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Peer is not supposed
to register
-
Regards,
AJ Stanfield
t: 0161-850-4001
e: a...@dmcip.com
w: http://www.dmcip.com
- Original Message -
From: Ishfaq Malik i...@pack-net.co.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, 4 November, 2013 3:36:06 PM
Subject: Re: [asterisk-users] No matching peers message has gone (1.8.23.1)
Hi
Thanks for the quick response. I'll read all the change logs from now on, I
promise!
Ish
On 4 November 2013 15:29, Joshua Colp jc...@digium.com wrote:
Ishfaq Malik wrote:
Hi
Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer
get the 'no matching peer' error when we get a dictionary SIP attack.
Now the logs always show a 'wrong password' when there actually isn't a
matching peer.
We even have alwaysauthreject = yes in our sip.conf.
Has anyone else noticed this phenomenon?
This is on purpose. To fix some exposure issues the code was changed to have an
internal peer (albeit one that can never successfully be authenticated against)
that gets used if no real peer is found. This reduces the chance (by a lot) of
the code exposing information in some off nominal cases.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
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 --
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asterisk-users mailing list
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--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET
LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
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Re: [asterisk-users] No matching peers message has gone (1.8.23.1)
Hi Arthur
It was a fail2ban based query and fail2ban is still working fine.
I was just trying to find out if the change was intentional or not.
Regards
Ish
On 4 November 2013 15:52, Arthur J. Stanfield a...@dmcip.com wrote:
Hi Ish,
I assume you are using Fail2Ban to monitor the logs for dictionary attacks
- If so, the following regex should work for 1.8:
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Wrong password
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - No matching
peer found
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Username/auth
name mismatch
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Device does not
match ACL
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Peer is not
supposed to register
-
Regards,
AJ Stanfield
t: 0161-850-4001
e: a...@dmcip.com
w: http://www.dmcip.com
- Original Message -
From: Ishfaq Malik i...@pack-net.co.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, 4 November, 2013 3:36:06 PM
Subject: Re: [asterisk-users] No matching peers message has gone (1.8.23.1)
Hi
Thanks for the quick response. I'll read all the change logs from now on,
I promise!
Ish
On 4 November 2013 15:29, Joshua Colp jc...@digium.com wrote:
Ishfaq Malik wrote:
Hi
Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer
get the 'no matching peer' error when we get a dictionary SIP attack.
Now the logs always show a 'wrong password' when there actually isn't a
matching peer.
We even have alwaysauthreject = yes in our sip.conf.
Has anyone else noticed this phenomenon?
This is on purpose. To fix some exposure issues the code was changed to
have an internal peer (albeit one that can never successfully be
authenticated against) that gets used if no real peer is found. This
reduces the chance (by a lot) of the code exposing information in some off
nominal cases.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
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:
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address:
PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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_
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asterisk-users mailing list
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--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
_
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[asterisk-users] CallerID settings
Hi all, What should I do when my E1/SIP provider need a specific callerid(num) setting for my outgoing calls? My problem is that if I use Set(CALLERID(num)=XXYY) then I got XXYY on the cdr src field, losing info of the real src of the call. What is the best way out? (Preferably tech independent) Thanks, Gabriel -- _ -- 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] Redirect a GSM call through Wifi to a SIP phone
Le 01/11/2013 18:02, A J Stiles a écrit : What you want is not just a SIP client; it also has to integrate with the phone's own GSM stack. You probably will have the most success with Android, because you are going to need well-documented Source Code to stand a chance of getting anywhere. You will need an Open Source SIP client and the Source Code for the stock Android GSM telephony app. You then will have to bodge the two together somehow . Actually, as long as the data packets coming from Android GSM phone are already compressed using the GSM codec, you shouldn't have to do anything to them to re-send them out over SIP; and similarly for packets arriving by the SIP connection to re-send over GSM. This really is not going to be a trivial project. Good luck with it -- you're going to need it :) I'm absolutely not developer, I wanted to know if it was possible. Thank you for your reply. Sil -- _ -- 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] Redirect a GSM call through Wifi to a SIP phone
Le 01/11/2013 18:54, ad...@3a.hu a écrit : Hi, On 11/1/2013 5:02 PM, A J Stiles wrote: You probably will have the most success with Android, because you are going to need well-documented Source Code to stand a chance of getting anywhere. You will need an Open Source SIP client and the Source Code for the stock Android GSM telephony app. You then will have to bodge the two together somehow . please correct me if I'm wrong, but if the above was possible, then also, current SIP softphone apps would be able to do - conference calls with a mixture of GSM and SIP endpoints - transfer calls from/to GSM to/from SIP which as far as I know, none of the apps offer. If I remember correctly, one of the softphone app programmers provided a reason why the above will never work: you can't access a GSM call from an app due to security reasons. It's true, I have not seen this problem. I made several tests with various softphones without be able to transfer a call on GSM to SIP. I'm not sure what percentage I'm right, but I'd strongly suggest a GSM gateway for the function either with ethernet and SIP support, or an FXO port. Much easier project. It was for transferring incoming calls on mobiles present in the office. From what I understand, a gateway is rather for outgoing calls. Thank you for your reply. Sil -- _ -- 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] CallerID settings
For outgoing calls you can write additional information into the userfield, or you can define your own additional fields using an adaptive-odbc setup. For ISDN and POTS channels you can typically set the callerid (just the number) for outgoing calls only to those numbers given to you by your telco (or they pick a default number). There are exceptions, but not for mere mortals. jg -- _ -- 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] two steps when calling from web!
Dear All. When I calling a number from web, my softphone show me Answer and Decline bottoms, and then I have to click Answer to call the number. it seems it is two step to calling the number. If I type the number direct to my client softphone, it calls directly the number without show me to choose Answer to calling. First call connect with client and then come into my screen and showing me to choose Answer and Decline.I'm not able to listen ringing sound because call is connecting first with client and then connect with my softphone. My source code is in AMI socket open to make call from web. how can I call direct to the number? -- _ -- 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
