Re: [asterisk-users] Call Limits
Would it not be easier for you to just bill them for access to 12 channels (6 extensions x 2 channels each)? Seems simpler. Then bill them for the calls they actually make. Then set "call-limit=2" for each "extension" in sip.conf? See: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Monday, 7 December 2009 00:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call Limits Hello, I'm trying to figure out how to limit the number of concurrent calls a client can make. I have a client that has 6 SIP accounts. One for each SIP phone. I want to limit it so that they can only make 2 outgoing calls at a time so that I can bill them "per channel" rather than "per extension". A separate (but not so important) issue is that I want them to be able to make unlimited numbers of calls without their 6 extensions. So that they can have a maximum of 2 outgoing calls, and the other 4 phones can still speak to each other without having to wait for one of the 2 other calls to end. I thought that maybe one way would be to duplicate the outbound sip settings and label them "outbound_client_1" and then use call-limit within that. Has anyone got any experience of this? Thanks Dan Journo Kesher Communications Ltd IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design & Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. ___ -- 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] [Asterisk-users] Error : SIP/2.0 401 Unauthorized
Hi Friends,
need to help.
*I have problem about sip : SIP/2.0 401 Unauthorized*
Is it require to nathelper module in kamailio ?
*what can i write kamailio.cfg file when kamailio and Asterisk on same
network?*
Scenario is like as :
-
1) kamailio server on 172.18.100.74
kamailio.cfg ( nathelpler module )
-
loadmodule "nathelper.so"
modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")
modparam("nathelper", "natping_interval", 60)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "rtpproxy_disable_tout", 60)
modparam("nathelper", "rtpproxy_tout", 1)
modparam("nathelper", "rtpproxy_retr", 5)
modparam("nathelper", "sipping_method", "OPTIONS")
modparam("nathelper", "received_avp", "$avp(i:801)")
2) Asterisk server on 172.18.100.65
sip.conf
---
[rajnikant]
nat=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
type=peer
context=default
host=172.18.100.74
fromdomain=rajnikant.net
mailbox=u...@context
--
Thanks and Regards
Rajnikant Vanza
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Re: [asterisk-users] Linksys SPA9x2 echo problem
On 7/12/09 8:11 AM, Dubravko Caric wrote: > Hi all, > we have > this annoying problem with Linksys SPA9x2 phones and echo cancellation. I > have read > posts on other sites about this problem but they are more than one year old > and > people were using older firmware. Linksys/Cisco has released 6.1.5a firmware > but we still experience the same problems. SPA phones have low sound volume of > handset microphone and people on the other side (PSTN, GSM) are complaining > that they hear us very badly. I increased Handset gain to +6 and lower > Additional > handset gain to -3 to get amplification of +3 what gives higher volume on the > other side but not as much as I would like. That is one problem, the other one > is that now VoIP to VoIP calls in our company are much louder than those to > PSTN and GSM, and if we receive two calls immediately one after another there > is a big difference between PSTN and VoIP calls in the sound volume on our > side > and our users are complaining. > We are > using SIP trunk towards PSTN (so I can’t use txgain and rxgain in > zapata.conf), > for GSM calls we are using PORTech SIPtoGSM GW (but I can’t increase volume on > it anymore because I am getting echo) and I can’t increase volume on Linksys > phones anymore because our internal VoIP calls would then be much too loud. > Does anyone > have same issues even with this new firmware and if so how do you handle > those problems? Sounds to me like you need to speak with the company providing you a SIP trunk. If the calls between VoIP->VoIP are too loud and the calls to PSTN are too quiet, then likely the provider needs to check their gain settings. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) ___ -- 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] realm
is the realm value used in authentication proccess is case sensitve?___ -- 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] How to use SIP hints and BLF for realtime extensions on Aastra phones?
Actually it was the 'call-limit' option which needed to have some value more than 0. By default the realtime extensions have it set to 0, and there was no such field in `sip-buddies` table. So I created it in `sip_buddies` and set it to 50, and now hints work just perfectly fine as they should. The only downside is that I have to enter static lines like "exten => 222,hint,SIP/222" for all the extensions under a separate context, as it looks like they can't be assigned as variables in extensions.ael or extensions.conf. I haven't tried putting them in the `extensions` table, but the documentation says hints don't work for contexts in `extensions` table. Any ideas? -- Zeeshan On Sun, Dec 6, 2009 at 1:53 PM, Philipp Kempgen wrote: > Zeeshan Zakaria schrieb: > > Actually BLF works fine on one system with Asterisk 1.4.22-4 but not on > the > > other one with Asterisk 1.4.18. Both have exact same sip.conf and > > extensions.conf, same extension numbers. Is there anything else which > could > > effect it. The one on which it doesn't work is on a virtual machine, on a > > virtual network. > > Try to narrow down the problem. I.e. try Asterisk 1.4.22-4 on > a virtual machine or Asterisk 1.4.18 on physical hardware. > > > On Sun, Dec 6, 2009 at 11:07 AM, Philipp Kempgen > > wrote: > > > >> Zeeshan Zakaria schrieb: > >> > I need to make use of BLF feature on Aastra 6757i phones but its an > >> Asterisk > >> > 1.4 using realtime architecture. Extensions > >> > >> "extensions" == sip.conf peers? > >> > >> > are defined in realtime database > >> > and dial plan is in AEL. I am able to correctly setup hints in the > >> dialplan, > >> > but they don't work. Did some research and found out that hints don't > >> work > >> > work with realtime extensions. > >> > >> They do, if rtcachefriends is enabled in sip.conf. > >> > >> > Is there any work around? > >> > > >> > On voip-info I read that Snom phones can use BLF without using hints. > >> > >> Huh? > >> > >> > Is it > >> > possible to do similar on Aastra phones? > >> > >> Carlos Chavez schrieb: > >> > You need to enable rtcachefriends=yes in sip.conf > >> > >> Zeeshan Zakaria schrieb: > >> > It is already enabled in sip.conf. > >> > >> All I can say is that "it should work". :-) > > >Philipp Kempgen > -- > AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de > Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 > Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de > Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de > -- > > ___ > -- 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 > -- Zeeshan A Zakaria ___ -- 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] only the first ResetCDR works after upgrade to 1.6
Andrew Witt wrote: > > I am upgrading from asterisk v1.2 to v1.6 and I am seeing a problem with > recording CDRs using MySQL. Unlike all of the other postings and web > pages I have found on this issue, my installation successfully stores > the -first- CDR, but nothing after that. It looks like the issues related to changes made in how NoCDR, ForkCDR, and ResetCDR in v1.4 work affect re-use of ResetCDR within a single phone call. e.g., bugs #12946, #13892, #16222 and others are related, even though they concern mainly v1.4 and call scenarios other than mine (such as transfers, non-answered calls, etc.) The bottom line is that the CDR status is "NO ANSWER" after calling ResetCDR the first time (even though the dialplan is still handling a call that was ANSWER'd.) By setting "unanswered = yes" in cdr.conf, multiple calls to ResetCDR during a single phone call produce output for each ResetCDR( ). -- Andrew Witt Sr. Software Systems Engineer revol wireless 216-525-1195 phone 216-240-1991 wireless 216-525-1112 fax andrew.w...@revol.com www.revol.com THIS MESSAGE IS INTENDED ONLY FOR THE USE OF THE INDIVIDUAL OR ENTITY TO WHICH IT IS ADDRESSED AND MAY CONTAIN INFORMATION THAT IS PRIVILEGED, CONFIDENTIAL, AND EXEMPT FROM DISCLOSURE UNDER APPLICABLE LAW. If the reader of this message is not the intended recipient, or the employee or agent responsible for delivering the message to the intended recipient, you are hereby notified that any dissemination, distribution, forwarding, or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by e-mail or telephone, and delete the original message immediately. ___ -- 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] Setting up skype
Julian Lyndon-Smith wrote: > Aha. That was it. Thanks. > > I could not see that advice in the documentation. I may be blind, but > it may be helpful to include it somewhere. Please open a ticket with Digium's support department documenting the issues you ran into setting up the product; that way it won't get lost and a future release will have improved documentation. Thanks! -- 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
[asterisk-users] Linksys SPA9x2 echo problem
Hi all, we have this annoying problem with Linksys SPA9x2 phones and echo cancellation. I have read posts on other sites about this problem but they are more than one year old and people were using older firmware. Linksys/Cisco has released 6.1.5a firmware but we still experience the same problems. SPA phones have low sound volume of handset microphone and people on the other side (PSTN, GSM) are complaining that they hear us very badly. I increased Handset gain to +6 and lower Additional handset gain to -3 to get amplification of +3 what gives higher volume on the other side but not as much as I would like. That is one problem, the other one is that now VoIP to VoIP calls in our company are much louder than those to PSTN and GSM, and if we receive two calls immediately one after another there is a big difference between PSTN and VoIP calls in the sound volume on our side and our users are complaining. We are using SIP trunk towards PSTN (so I can’t use txgain and rxgain in zapata.conf), for GSM calls we are using PORTech SIPtoGSM GW (but I can’t increase volume on it anymore because I am getting echo) and I can’t increase volume on Linksys phones anymore because our internal VoIP calls would then be much too loud. Does anyone have same issues even with this new firmware and if so how do you handle those problems? Any help would be much appreciated Best regards Dubravko ___ -- 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] Asterisk to Email
On Sun, Dec 6, 2009 at 12:28 PM, Thomas Perron wrote:
> I am reading a lot of the material but need your input to help me
> understand what you mean.
>
> System(echo body of message | mail -s "subject line"
> ${the_caller_...@tmobile.net)
>
> I understand the System application generally
> echo body of message .?
> mail -s --what does this do please?
> "subject line" .comes from where?
> ${the_caller_...@tmobile.net) i understand this part.
>
> thank you
>
>
>
>
> On Sun, Dec 6, 2009 at 2:40 AM, Tzafrir Cohen
> wrote:
>> On Sat, Dec 05, 2009 at 08:25:33PM -0500, Thomas Perron wrote:
>>
>>> And, then send an email to the party. Example
>>>
>>> 3035551...@tmobile.net
>>>
>>> Summary
>>> 1. Capture the CID number.
>>> 2. Prepend his number to his service provider SMTP address
>>> 3. Email it to his phone
>>
>>
>> System(echo body of message | mail -s "subject line"
>> ${the_caller_...@tmobile.net)
>>
>> Note the usage of '|' here. IIRC it needs to be escaped on Asterisk
>> 1.4.x and below.
>>
>>>
>>> I assume I need to install SendMail and play around with CID stuff.
>>
>> Sendmail, postfix, exim, qmail - any program that provides a local
>> sendmail interface.
>>
>> I personally prefer postfix.
>>
>> --
>> Tzafrir Cohen
>> icq#16849755 jabber:tzafrir.co...@xorcom.com
>> +972-50-7952406 mailto:tzafrir.co...@xorcom.com
>> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
>>
>> ___
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
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>
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>
What everybody is trying to tell you is you need get this working from
the command line before you do it in asterisk.
The hardest part will probably be setting up your mail application. I
found postfix the easiest to set up, but YMMV. In any event you've got
to be able to use "mail" from the command line. Once you do that, then
try using it with System() in asterisk.
sean
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Re: [asterisk-users] How to use SIP hints and BLF for realtime extensions on Aastra phones?
Zeeshan Zakaria schrieb: > Actually BLF works fine on one system with Asterisk 1.4.22-4 but not on the > other one with Asterisk 1.4.18. Both have exact same sip.conf and > extensions.conf, same extension numbers. Is there anything else which could > effect it. The one on which it doesn't work is on a virtual machine, on a > virtual network. Try to narrow down the problem. I.e. try Asterisk 1.4.22-4 on a virtual machine or Asterisk 1.4.18 on physical hardware. > On Sun, Dec 6, 2009 at 11:07 AM, Philipp Kempgen > wrote: > >> Zeeshan Zakaria schrieb: >> > I need to make use of BLF feature on Aastra 6757i phones but its an >> Asterisk >> > 1.4 using realtime architecture. Extensions >> >> "extensions" == sip.conf peers? >> >> > are defined in realtime database >> > and dial plan is in AEL. I am able to correctly setup hints in the >> dialplan, >> > but they don't work. Did some research and found out that hints don't >> work >> > work with realtime extensions. >> >> They do, if rtcachefriends is enabled in sip.conf. >> >> > Is there any work around? >> > >> > On voip-info I read that Snom phones can use BLF without using hints. >> >> Huh? >> >> > Is it >> > possible to do similar on Aastra phones? >> >> Carlos Chavez schrieb: >> > You need to enable rtcachefriends=yes in sip.conf >> >> Zeeshan Zakaria schrieb: >> > It is already enabled in sip.conf. >> >> All I can say is that "it should work". :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de -- ___ -- 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] Setting up skype
Aha. That was it. Thanks. I could not see that advice in the documentation. I may be blind, but it may be helpful to include it somewhere. Thanks again Julian 2009/12/6 Kevin P. Fleming : > Julian Lyndon-Smith wrote: >> That's my point - SFA comes with a g729 licence, so why can't it >> transcode to the DAHDI channel ? > > It comes with a license, but does not include the transcoding > functionality itself. You need to download and install the appropriate > Digium codec_g729 module for your system to enable transcoding using > that license. > > -- > 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] Asterisk to Email
Thomas Perron schrieb:
> I am reading a lot of the material but need your input to help me
> understand what you mean.
>
> System(echo body of message | mail -s "subject line"
> ${the_caller_...@tmobile.net)
>
> I understand the System application generally
> echo body of message .?
Please read the man page of the `echo` command.
man echo
( http://unixhelp.ed.ac.uk/CGI/man-cgi?echo )
> mail -s --what does this do please?
> "subject line" .comes from where?
man mail
( http://unixhelp.ed.ac.uk/CGI/man-cgi?mail )
> ${the_caller_...@tmobile.net) i understand this part.
> On Sun, Dec 6, 2009 at 2:40 AM, Tzafrir Cohen
> wrote:
>> On Sat, Dec 05, 2009 at 08:25:33PM -0500, Thomas Perron wrote:
>>
>>> And, then send an email to the party. Example
>>>
>>> 3035551...@tmobile.net
>>>
>>> Summary
>>> 1. Capture the CID number.
>>> 2. Prepend his number to his service provider SMTP address
>>> 3. Email it to his phone
>>
>>
>> System(echo body of message | mail -s "subject line"
>> ${the_caller_...@tmobile.net)
>>
>> Note the usage of '|' here. IIRC it needs to be escaped on Asterisk
>> 1.4.x and below.
>>
>>>
>>> I assume I need to install SendMail and play around with CID stuff.
>>
>> Sendmail, postfix, exim, qmail - any program that provides a local
>> sendmail interface.
>>
>> I personally prefer postfix.
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de
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Re: [asterisk-users] Asterisk to Email
Try Googling some of this stuff, such as linux "mail -s" --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron mail -s --what does this do please? ___ -- 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] How to use SIP hints and BLF for realtime extensions on Aastra phones?
Actually BLF works fine on one system with Asterisk 1.4.22-4 but not on the other one with Asterisk 1.4.18. Both have exact same sip.conf and extensions.conf, same extension numbers. Is there anything else which could effect it. The one on which it doesn't work is on a virtual machine, on a virtual network. -- Zeeshan On Sun, Dec 6, 2009 at 11:07 AM, Philipp Kempgen wrote: > Zeeshan Zakaria schrieb: > > I need to make use of BLF feature on Aastra 6757i phones but its an > Asterisk > > 1.4 using realtime architecture. Extensions > > "extensions" == sip.conf peers? > > > are defined in realtime database > > and dial plan is in AEL. I am able to correctly setup hints in the > dialplan, > > but they don't work. Did some research and found out that hints don't > work > > work with realtime extensions. > > They do, if rtcachefriends is enabled in sip.conf. > > > Is there any work around? > > > > On voip-info I read that Snom phones can use BLF without using hints. > > Huh? > > > Is it > > possible to do similar on Aastra phones? > > Carlos Chavez schrieb: > > You need to enable rtcachefriends=yes in sip.conf > > Zeeshan Zakaria schrieb: > > It is already enabled in sip.conf. > > All I can say is that "it should work". :-) > > >Philipp Kempgen > -- > AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de > Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 > Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de > Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de > -- > > ___ > -- 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 > -- Zeeshan A Zakaria ___ -- 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] Asterisk to Email
I am reading a lot of the material but need your input to help me
understand what you mean.
System(echo body of message | mail -s "subject line"
${the_caller_...@tmobile.net)
I understand the System application generally
echo body of message .?
mail -s --what does this do please?
"subject line" .comes from where?
${the_caller_...@tmobile.net) i understand this part.
thank you
On Sun, Dec 6, 2009 at 2:40 AM, Tzafrir Cohen wrote:
> On Sat, Dec 05, 2009 at 08:25:33PM -0500, Thomas Perron wrote:
>
>> And, then send an email to the party. Example
>>
>> 3035551...@tmobile.net
>>
>> Summary
>> 1. Capture the CID number.
>> 2. Prepend his number to his service provider SMTP address
>> 3. Email it to his phone
>
>
> System(echo body of message | mail -s "subject line"
> ${the_caller_...@tmobile.net)
>
> Note the usage of '|' here. IIRC it needs to be escaped on Asterisk
> 1.4.x and below.
>
>>
>> I assume I need to install SendMail and play around with CID stuff.
>
> Sendmail, postfix, exim, qmail - any program that provides a local
> sendmail interface.
>
> I personally prefer postfix.
>
> --
> Tzafrir Cohen
> icq#16849755 jabber:tzafrir.co...@xorcom.com
> +972-50-7952406 mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] ABCTI: first usable beta
Hallo, ABCTI (an open-source CTI client for Asterisk) has moved to beta stage. Find it on: http://abcti.sourceforge.net For the first time, we now have windows installers that actually work ;-) We would appreciate any feedback you can give. Regards, -- 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] Can Asterik act as a SIP Proxy
Gayathri G schrieb: > 1. Can I use Asterisk as a SIP Proxy. ( I want it to act as proxy not a > B2b/GW) No. Asterisk is a back-to-back user agent (B2BUA). You might want to have a look at http://en.wikipedia.org/wiki/OpenSER Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de -- ___ -- 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] How to use SIP hints and BLF for realtime extensions on Aastra phones?
Zeeshan Zakaria schrieb: > I need to make use of BLF feature on Aastra 6757i phones but its an Asterisk > 1.4 using realtime architecture. Extensions "extensions" == sip.conf peers? > are defined in realtime database > and dial plan is in AEL. I am able to correctly setup hints in the dialplan, > but they don't work. Did some research and found out that hints don't work > work with realtime extensions. They do, if rtcachefriends is enabled in sip.conf. > Is there any work around? > > On voip-info I read that Snom phones can use BLF without using hints. Huh? > Is it > possible to do similar on Aastra phones? Carlos Chavez schrieb: > You need to enable rtcachefriends=yes in sip.conf Zeeshan Zakaria schrieb: > It is already enabled in sip.conf. All I can say is that "it should work". :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de -- ___ -- 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] sequential dialing preferences
I am trying to use a simple tool in the Dial plan so that if the first number does not connect the logic will go to the second and/or third. Basically, I want the call to ring and connect to the first number Then, if it is not answered I want another number to try to get connected Then, if second number does not answer I want the third to be tried i only list the scenario for the first two numbers Here is what I have now which works fine for the one and only number... exten => s,n,Dial(SIP/callwithus/12135551212,120,A(ginger3)) ; Service line so, will this work ... .. exten => s,n,Dial(SIP/callwithus/12135551212[&SIP/callwithus/12145551212],120,A(ginger3)) ; Service line Please send comments to make this work. Thanks ___ -- 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] Setting up skype
Julian Lyndon-Smith wrote: > That's my point - SFA comes with a g729 licence, so why can't it > transcode to the DAHDI channel ? It comes with a license, but does not include the transcoding functionality itself. You need to download and install the appropriate Digium codec_g729 module for your system to enable transcoding using that license. -- 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
[asterisk-users] Call Limits
Hello, I'm trying to figure out how to limit the number of concurrent calls a client can make. I have a client that has 6 SIP accounts. One for each SIP phone. I want to limit it so that they can only make 2 outgoing calls at a time so that I can bill them "per channel" rather than "per extension". A separate (but not so important) issue is that I want them to be able to make unlimited numbers of calls without their 6 extensions. So that they can have a maximum of 2 outgoing calls, and the other 4 phones can still speak to each other without having to wait for one of the 2 other calls to end. I thought that maybe one way would be to duplicate the outbound sip settings and label them "outbound_client_1" and then use call-limit within that. Has anyone got any experience of this? Thanks Dan Journo Kesher Communications Ltd ___ -- 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] Asterisk to Email
Thomas Perron wrote:
> Interesting response but I am not that saavy to follow it!
> Thank you
>
>
>
If you can't follow his response, perhaps you need to hire someone to
write your dialplan?
I know next to nothing regarding coding, and nothing of 1.6, and I
understood it.
He practically wrote it for you!
Start reading!!
Moon Mullins
> On Sun, Dec 6, 2009 at 2:40 AM, Tzafrir Cohen
> wrote:
>
>> On Sat, Dec 05, 2009 at 08:25:33PM -0500, Thomas Perron wrote:
>>
>>
>>> And, then send an email to the party. Example
>>>
>>> 3035551...@tmobile.net
>>>
>>> Summary
>>> 1. Capture the CID number.
>>> 2. Prepend his number to his service provider SMTP address
>>> 3. Email it to his phone
>>>
>> System(echo body of message | mail -s "subject line"
>> ${the_caller_...@tmobile.net)
>>
>> Note the usage of '|' here. IIRC it needs to be escaped on Asterisk
>> 1.4.x and below.
>>
>>
>>> I assume I need to install SendMail and play around with CID stuff.
>>>
>> Sendmail, postfix, exim, qmail - any program that provides a local
>> sendmail interface.
>>
>> I personally prefer postfix.
>>
>> --
>> Tzafrir Cohen
>> icq#16849755 jabber:tzafrir.co...@xorcom.com
>> +972-50-7952406 mailto:tzafrir.co...@xorcom.com
>> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
>>
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>>
>
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Re: [asterisk-users] Asterisk to Email
Thomas Perron wrote: > Interesting response but I am not that saavy to follow it! > Thank you > Then I'd suggest you hire someone that is more technically inclined to help you with your project. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- 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] Asterisk to Email
Interesting response but I am not that saavy to follow it!
Thank you
On Sun, Dec 6, 2009 at 2:40 AM, Tzafrir Cohen wrote:
> On Sat, Dec 05, 2009 at 08:25:33PM -0500, Thomas Perron wrote:
>
>> And, then send an email to the party. Example
>>
>> 3035551...@tmobile.net
>>
>> Summary
>> 1. Capture the CID number.
>> 2. Prepend his number to his service provider SMTP address
>> 3. Email it to his phone
>
>
> System(echo body of message | mail -s "subject line"
> ${the_caller_...@tmobile.net)
>
> Note the usage of '|' here. IIRC it needs to be escaped on Asterisk
> 1.4.x and below.
>
>>
>> I assume I need to install SendMail and play around with CID stuff.
>
> Sendmail, postfix, exim, qmail - any program that provides a local
> sendmail interface.
>
> I personally prefer postfix.
>
> --
> Tzafrir Cohen
> icq#16849755 jabber:tzafrir.co...@xorcom.com
> +972-50-7952406 mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
>
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Re: [asterisk-users] Example to handle incoming calls without callerid at home?
James you are right. let me add one more line
exten => s,n,GotoIf($[${CALLERID(num)}=""]?nocid,s,1)
On Sun, Dec 6, 2009 at 3:18 PM, James Stocks wrote:
> On 6 Dec 2009, at 08:56, Remco Barendse wrote:
>
> > I am using asterisk 1.6 at home and would like to send incoming calls
> > without caller id immediately to voicemail (i don't want to use the
> > privacy manager where people have to enter a number).
> >
> > The config examples i found are all for the pretty obsolete 1.0 and 1.2
> > versions of asterisk.
> >
> > Would anyone be willing to share a config example?
> >
> > Thanks!
>
> Well I don't claim to be a guru, but this is what I do:
>
> ; This is the context which receives calls:
> [from-pstn]
> exten => s,1,GotoIf($["${CALLERID(name)}" = "WITHHELD"]?nocid,s,1)
> exten => s,n,GotoIf($["${CALLERID(name)}" = "INTERNATIONAL"]?nocid,s,1)
> exten => s,n,GotoIf($["${CALLERID(name)}" = "UNAVAILABLE"]?nocid,s,1)
> exten => s,n,GotoIf($["${CALLERID(name)}" = "PAYPHONE"]?nocid,s,1)
> exten => s,n,Macro(call-house-phones)
> exten => s,n,Hangup
>
> [nocid]
> ; If no caller ID, here's where you specify what to do:
> exten => s,1,Voicemail(401,u)
> exten => s,n,Hangup
>
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Re: [asterisk-users] Example to handle incoming calls without callerid at home?
On 6 Dec 2009, at 08:56, Remco Barendse wrote:
> I am using asterisk 1.6 at home and would like to send incoming calls
> without caller id immediately to voicemail (i don't want to use the
> privacy manager where people have to enter a number).
>
> The config examples i found are all for the pretty obsolete 1.0 and 1.2
> versions of asterisk.
>
> Would anyone be willing to share a config example?
>
> Thanks!
Well I don't claim to be a guru, but this is what I do:
; This is the context which receives calls:
[from-pstn]
exten => s,1,GotoIf($["${CALLERID(name)}" = "WITHHELD"]?nocid,s,1)
exten => s,n,GotoIf($["${CALLERID(name)}" = "INTERNATIONAL"]?nocid,s,1)
exten => s,n,GotoIf($["${CALLERID(name)}" = "UNAVAILABLE"]?nocid,s,1)
exten => s,n,GotoIf($["${CALLERID(name)}" = "PAYPHONE"]?nocid,s,1)
exten => s,n,Macro(call-house-phones)
exten => s,n,Hangup
[nocid]
; If no caller ID, here's where you specify what to do:
exten => s,1,Voicemail(401,u)
exten => s,n,Hangup
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Re: [asterisk-users] Setting up skype
That's my point - SFA comes with a g729 licence, so why can't it transcode to the DAHDI channel ? Thanks also for the info. Very useful. Julian 2009/12/6 Roeften : > From what I understand your sip client can handle g729 whereas for DAHDI you > need transcoding to a|ulaw. > > I am using it with no problems (have g729 licenses as well though). > > A bit off topic, I have found some extra configuration that is not really in > the docs (or I could not find them): > > fullname=Your full name > country=gr > language=en > city=City > province=Province > phone_home=+fullinternationalnumber > phone_office=+fullinternationalnumber > email=y...@email.com > homepage=http://www.example.com > avatar=/var/lib/asterisk/images/skype100x100.jpg > > Just a note the country code has to be lower case (i.e GR would not work). > > Panos > > On Sun, Dec 6, 2009 at 9:40 AM, Julian Lyndon-Smith > wrote: >> >> Ok. So I bought 2x skpye channels. Doesn't that mean I have 2xg729 as well >> ? >> >> If so, why do I have the problem ? And would this affect local >> channels as well ? >> >> Julian >> >> 2009/12/6 Kevin P. Fleming : >> > Julian Lyndon-Smith wrote: >> > >> >> external => ddi => dial(skype) >> >> >> >> and got a load of static with >> >> >> >> WARNING[15328]: channel.c:3098 set_format: Unable to find a codec >> >> translation path from 0x100 (g729) to 0x8 (alaw) >> >> >> >> on the console. >> >> >> >> Fired up a sip client, made the same call, and all was ok. >> >> >> >> Any clues ? >> > >> > The clues are in the documentation; SkypeIn and SkypeOut use G.729 for >> > nearly all calls, so handling calls via those paths requires a G.729 >> > transcoder on the system if the target of the call will not also be >> > using G.729. This is why the Skype For Asterisk license includes >> > licenses for Digium's G.729 software transcoder as well. >> > >> > -- >> > 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 > > > ___ > -- 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] Example to handle incoming calls without callerid at home?
I am using asterisk 1.6 at home and would like to send incoming calls without caller id immediately to voicemail (i don't want to use the privacy manager where people have to enter a number). The config examples i found are all for the pretty obsolete 1.0 and 1.2 versions of asterisk. Would anyone be willing to share a config example? Thanks! ___ -- 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] Setting up skype
>From what I understand your sip client can handle g729 whereas for DAHDI you need transcoding to a|ulaw. I am using it with no problems (have g729 licenses as well though). A bit off topic, I have found some extra configuration that is not really in the docs (or I could not find them): fullname=Your full name country=gr language=en city=City province=Province phone_home=+fullinternationalnumber phone_office=+fullinternationalnumber email=y...@email.com homepage=http://www.example.com avatar=/var/lib/asterisk/images/skype100x100.jpg Just a note the country code has to be lower case (i.e GR would not work). Panos On Sun, Dec 6, 2009 at 9:40 AM, Julian Lyndon-Smith wrote: > Ok. So I bought 2x skpye channels. Doesn't that mean I have 2xg729 as well > ? > > If so, why do I have the problem ? And would this affect local > channels as well ? > > Julian > > 2009/12/6 Kevin P. Fleming : > > Julian Lyndon-Smith wrote: > > > >> external => ddi => dial(skype) > >> > >> and got a load of static with > >> > >> WARNING[15328]: channel.c:3098 set_format: Unable to find a codec > >> translation path from 0x100 (g729) to 0x8 (alaw) > >> > >> on the console. > >> > >> Fired up a sip client, made the same call, and all was ok. > >> > >> Any clues ? > > > > The clues are in the documentation; SkypeIn and SkypeOut use G.729 for > > nearly all calls, so handling calls via those paths requires a G.729 > > transcoder on the system if the target of the call will not also be > > using G.729. This is why the Skype For Asterisk license includes > > licenses for Digium's G.729 software transcoder as well. > > > > -- > > 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 > ___ -- 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
