[asterisk-users] Moderated News Aggregation for Asterisk
Hi all. Just wanted to let people know about a small project I started over the weekend to help me keep up with news about Asterisk. http://asterisktimes.xdev.net/ Some of the other new sites are either not there anymore or slow to update, so I've come up with a different idea for keeping Asterisk news up-to date and in one place. For the moment, I call it Asterisk Times. OK, so maybe not the best name, but it's a work in progress. So what is it? Well, this is an attempt to create a moderated, aggregated news platform for Asterisk. We want developers, 3rd party companies, open source tools, in fact anyone who does anything noteworthy with Asterisk to tell us. And the best way to do it, is by letting us have an RSS feed into your own announcements or news. With that, we can then review and submit news to the aggregator on your behalf, which then shows up on the homepage of this website. I hope people see value in this, as I know I do. This isn't run for profit or commercial reasons, it's just because I think as a community we deserve a better, more frequently updated news site. The URL will change (or at least get its own dedicated URL) once the project is off the ground and I can see people getting value from it. Any suggestions or feedback welcome. Thanks, Ben (aka skrusty on irc) -- _ -- 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] DTLS setting impacts encryption setting
If I understand correctly, setting encryption=no means that Asterisk will make outgoing calls without encryption, but will be happy to accept incoming calls regardless of whether the caller wants encryption or not If encryption=yes, then Asterisk not only uses encryption for the outgoing calls but it will refuse to accept incoming calls unless they use encryption too If I have encryption=no dtlsenable=yes the DTLS support works but Asterisk will no longer accept incoming calls using regular RTP/AVP. These messages appear in the console and the call is rejected with code 488: [Jan 28 11:08:42] WARNING[24673][C-0009]: chan_sip.c:10496 process_sdp: Processed DTLS [FALSE] [Jan 28 11:08:42] WARNING[24673][C-0009]: chan_sip.c:10529 process_sdp: We are requesting SRTP for audio, but they responded without it! I realise not everybody would set encryption=no in this situation, I'm simply trying to make it work for all possible callers to the SIP5060.net test numbers at http://www.sip5060.net/test-calls -- _ -- 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] grp_lock error when compiling against pjproject
On Tue, Jan 28, 2014 at 2:40 AM, Ira i...@extrasensory.com wrote: Hello Matthew, Monday, January 27, 2014, 1:49:44 PM, you wrote: Do you have the exact error message that pjproject gave when you ran into this problem? I don't, but I guess I can reinstall the offending software to get it if you need it. It's documented on the bug list as I eventually found the using google. I'm including the asterisk-users mailing list on this reply, as there's no reason to take this discussion off list. I'm not sure what bug list you're referring to. However, the page on the wiki that documents common errors and their appropriate correction [1] attempts to provide the exact error message that users will see when they encounter that situation. I'd be happy to update it with whatever error you ran into, but to do so we need to know the exact messages. Alluding to error messages without providing them usually leads to more confusion, not less. [1] https://wiki.asterisk.org/wiki/display/AST/Installing+pjproject#Installingpjproject-IssuesandWorkarounds Matt -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dimensioning
I have been trying to get a feel for scaling or dimensioning using asterisk 11. if I desire to use something like a dell r320, hardware RAID, 2G E5-2420, 4G RAM and only SIP trunking using gsm (least bandwidth and no transcoding) how many calls out can I expect to make at one time and asterisk still be OK and responsive? Thanks, Jerry -- _ -- 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] dimensioning
On 28/01/14 15:01, Jerry Geis wrote: I have been trying to get a feel for scaling or dimensioning using asterisk 11. if I desire to use something like a dell r320, hardware RAID, 2G E5-2420, 4G RAM and only SIP trunking using gsm (least bandwidth and no transcoding) how many calls out can I expect to make at one time and asterisk still be OK and responsive? Thanks, Jerry We have a quad 'Intel(R) Xeon(R) CPU E5-1410 0 @ 2.80GHz' that will copy with about 800 concurrent calls with a lot of AGI stuff the log call start and end and about 5-10% of the calls being recorded. All using g711alaw. All calls come in and go out via sip. -- _ -- 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] AMI eventmask question
On Thu, Jan 23, 2014 at 9:05 PM, Daniel Jenkins dan.jenkin...@gmail.comwrote: On Thu, Jan 23, 2014 at 8:46 PM, Matthew Jordan mjor...@digium.comwrote: On Thu, Jan 23, 2014 at 9:31 AM, Michelle Dupuis mdup...@ocg.ca wrote: Thanks - I've been through that doc before and couldn't find the info needed, which is why I went to the source code eventually. All events are grouped, and each group is given a name/flag like 'system', 'call', etc. The docs just don't say which events are in which group/flag. Perhaps something Digium could add at some point :) Or someone from the open source community... this is an open source project, after all :-) The managerEventInstance XML elements already have an attribute for the manager class, which is populated: managerEvent language=en_US name=ParkedCallTimeOut managerEventInstance class=EVENT_FLAG_CALL synopsisRaised when a parked call times out./synopsis syntax parameter name=Exten paraThe parking lot extension./para /parameter parameter name=Channel/ parameter name=Parkinglot paraThe name of the parking lot./para /parameter parameter name=CallerIDNum/ parameter name=CallerIDName/ parameter name=ConnectedLineNum/ parameter name=ConnectedLineName/ parameter name=UniqueID/ /syntax see-also ref type=managerEventParkedCall/ref /see-also /managerEventInstance /managerEvent You could actually grep the core-en_US.xml file and get all of the events that match to a particular class authorization. It doesn't show up in the CLI due to the xmldoc API not parsing out that attribute. The same is true for the wiki documentation; that project is up on github [1]. It wouldn't be a large patch to either to have that attribute displayed. Matt [1] https://github.com/asterisk/publish-docs -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks Matt, I was going to ask where the tool was that generated from source, I'll take a look and see if I can contribute that back, Dan Hi, just to let you know that this is partially done now, https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerEvent_AgentConnect At the bottom you'll see a Class Next stage is to create a page which has all the classes and the events grouped by class. But at least you can see which event is what class now. Dan -- _ -- 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] [HELP]: Auto-answering calls placed from call files
Hello All, I've asked this on the asterisk-dev list, so sorry for cross-posting. So far I'm not sure how to accomplish this without looking at the source code or looking at some other way to get around this issue. I'm trying to have an automated call to an Aastra SIP phone and have the call auto-answeredby the phone. I know that a SIP call placed to the phone can be auto-answered if a certain SIP header is added to the call. I am able to apply the SIP headers manually and get that working (using SIPAddHeader(Alert-Info: info=alert-autoanswer) in the dialplan, but for call files, I don't seem to be able to edit any of the sip headers - there is only basic customizations allowed to setup the calls. Does anyone know how I could place automated outgoing calls that would have the proper sip headers added to it that would allow the call to be auto-answered? I've also posted this question to the forums here: http://forums.asterisk.org/viewtopic.php?f=1t=89190 Many thanks, Steve http://www.stevemccann.net -- _ -- 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] grp_lock error when compiling against pjproject
On Tue, Jan 28, 2014 at 6:29 AM, Matthew Jordan mjor...@digium.com wrote: On Tue, Jan 28, 2014 at 2:40 AM, Ira i...@extrasensory.com wrote: Hello Matthew, Monday, January 27, 2014, 1:49:44 PM, you wrote: Do you have the exact error message that pjproject gave when you ran into this problem? I don't, but I guess I can reinstall the offending software to get it if you need it. It's documented on the bug list as I eventually found the using google. I'm including the asterisk-users mailing list on this reply, as there's no reason to take this discussion off list. I'm not sure what bug list you're referring to. However, the page on the wiki that documents common errors and their appropriate correction [1] attempts to provide the exact error message that users will see when they encounter that situation. I'd be happy to update it with whatever error you ran into, but to do so we need to know the exact messages. Alluding to error messages without providing them usually leads to more confusion, not less. I ran into the same problem a few weeks ago. The error is during the asterisk-12 build and it's related to using a version of pjproject from before file made a grp_lock commit in pjproject in September. res_pjsip/pjsip_distributor.c: In function ‘find_dialog’: res_pjsip/pjsip_distributor.c:139:25: error: ‘pjsip_transaction’ has no member named ‘grp_lock’ pj_grp_lock_release(tsx-grp_lock); ^ make[1]: *** [res_pjsip/pjsip_distributor.o] Error 1 The fix is to clean out all old copies pf pjproject and re-clone from github. -- _ -- 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] [HELP]: Auto-answering calls placed from call files
On 28/01/14 16:56, Steve McCann wrote: Hello All, I've asked this on the asterisk-dev list, so sorry for cross-posting. So far I'm not sure how to accomplish this without looking at the source code or looking at some other way to get around this issue. I'm trying to have an automated call to an Aastra SIP phone and have the call auto-answeredby the phone. I know that a SIP call placed to the phone can be auto-answered if a certain SIP header is added to the call. I am able to apply the SIP headers manually and get that working (using SIPAddHeader(Alert-Info: info=alert-autoanswer) in the dialplan, but for call files, I don’t seem to be able to edit any of the sip headers - there is only basic customizations allowed to setup the calls. Does anyone know how I could place automated outgoing calls that would have the proper sip headers added to it that would allow the call to be auto-answered? I've also posted this question to the forums here: http://forums.asterisk.org/viewtopic.php?f=1t=89190 http://forums.asterisk.org/viewtopic.php?f=1t=89190 Many thanks, Steve So I take it in the call file you have it set to call Dial(SIP/something) ? If rather than dialling the sip destination immediately you dialled a local channel then it could add the custom header and then initiate the dial to the sip destination. -- _ -- 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] [HELP]: Auto-answering calls placed from call files
On Tue, Jan 28, 2014 at 10:56 AM, Steve McCann srmcc...@gmail.com wrote: Hello All, I've asked this on the asterisk-dev list, so sorry for cross-posting. So far I'm not sure how to accomplish this without looking at the source code or looking at some other way to get around this issue. I'm trying to have an automated call to an Aastra SIP phone and have the call auto-answeredby the phone. I know that a SIP call placed to the phone can be auto-answered if a certain SIP header is added to the call. I am able to apply the SIP headers manually and get that working (using SIPAddHeader(Alert-Info: info=alert-autoanswer) in the dialplan, but for call files, I don't seem to be able to edit any of the sip headers - there is only basic customizations allowed to setup the calls. Does anyone know how I could place automated outgoing calls that would have the proper sip headers added to it that would allow the call to be auto-answered? I've also posted this question to the forums here: http://forums.asterisk.org/viewtopic.php?f=1t=89190 Many thanks, Steve This isn't a development question, as it doesn't relate to the actual Asterisk source code itself. Cross-posting across the -dev and -users lists isn't helpful either, as pretty much everyone who is subscribed to the asterisk-dev list is also subscribed to the asterisk-users list. As SIPAddHeader is a dialplan application and not a dialplan function, it cannot be used from a call file. One approach to performing an outbound call that requires SIPAddHeader - and that doesn't rely on undocumented behaviour - is to use the call file to create a Local channel in the dialplan that dials the SIP channel, and use SIPAddHeader from there. A quick Google indicates others have used a similar approach in the past as well [1]. [1] http://lists.digium.com/pipermail/asterisk-users/2008-January/204375.html Matt -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [HELP]: Auto-answering calls placed from call files
Although I haven't tried this for this particular example, instead of using a .call file, you could probably originate a call using Ryan Bullock's Asterisk::AMI PERL module http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.8. It's one of the most valuable tools that I have and I've written literally hundreds of PERL scripts using it. You should check it out. It's got good documentation and examples to go along with it. I also use the AGISpeedy FastAGI package written in PERL and there's also an AGISpeedy package written in php that is also a valuable tool. Regards; John -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan Sent: Tuesday, January 28, 2014 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [HELP]: Auto-answering calls placed from call files On Tue, Jan 28, 2014 at 10:56 AM, Steve McCann srmcc...@gmail.com wrote: Hello All, I've asked this on the asterisk-dev list, so sorry for cross-posting. So far I'm not sure how to accomplish this without looking at the source code or looking at some other way to get around this issue. I'm trying to have an automated call to an Aastra SIP phone and have the call auto-answeredby the phone. I know that a SIP call placed to the phone can be auto-answered if a certain SIP header is added to the call. I am able to apply the SIP headers manually and get that working (using SIPAddHeader(Alert-Info: info=alert-autoanswer) in the dialplan, but for call files, I don't seem to be able to edit any of the sip headers - there is only basic customizations allowed to setup the calls. Does anyone know how I could place automated outgoing calls that would have the proper sip headers added to it that would allow the call to be auto-answered? I've also posted this question to the forums here: http://forums.asterisk.org/viewtopic.php?f=1t=89190 Many thanks, Steve This isn't a development question, as it doesn't relate to the actual Asterisk source code itself. Cross-posting across the -dev and -users lists isn't helpful either, as pretty much everyone who is subscribed to the asterisk-dev list is also subscribed to the asterisk-users list. As SIPAddHeader is a dialplan application and not a dialplan function, it cannot be used from a call file. One approach to performing an outbound call that requires SIPAddHeader - and that doesn't rely on undocumented behaviour - is to use the call file to create a Local channel in the dialplan that dials the SIP channel, and use SIPAddHeader from there. A quick Google indicates others have used a similar approach in the past as well [1]. [1] http://lists.digium.com/pipermail/asterisk-users/2008-January/204375.html Matt -- 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 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] B410P BRI card testing with patgen / pattest
Hello, I'm not sure if my previous e-mail went through, it contained a link so it may have been blocked, Apologies if this is a duplicate. I have a card here at the office that I'm trying to validate with some tests. For now I've followed the instructions available in the KB (Back-to-Back-Pattern-Test-for-BRI-Adapters) I get this kind of output: [..] (.Error 44119): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44120): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44121): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44122): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44123): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44124): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44125): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44126): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44127): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44128): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44129): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44130): Unexpected result, 255 != 0, 1 bytes since last error. [...] This goes on continuously, like a massive flood, not a small burst in the beginning like the KB article mentions. Would this be the kind of output that matches a problem card or something may be wrong in my setup? I'm using DAHDI 2.8.0.1. Any input appreciated, thanks! Regards, Rodrigo -- _ -- 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] callerid overwrite
Hi all,
I'm having issues with overwrite caller id, when I call someone my caller
id should be mycompanyinc but instead my id shows up as my extension
number 101.
this is what i have in sip.conf
[101]
type=friend
context=sipphones
call-limit=99
callerid=iuser 101
disallow=all
allow=ulaw
allow=alaw
username=101
secret=Passwd
dtmfmode=rfc2833
host=dynamic
mailbox=101@default
nat=yes
canreinvite=no
this is what i have in extensions.conf
[outbound]
exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)
exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)
any ideas? as this happens random,
--
_
-- 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] Integration with outlook
Hello; Is there a method way to be able to dial the phone number through asterisk from the outlook email contact? Regards Bilal-- _ -- 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] Integration with outlook
Hi Yes, there is, I am using http://outcall.sourceforge.net/ it's opensource. On Tue, Jan 28, 2014 at 2:13 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; Is there a method way to be able to dial the phone number through asterisk from the outlook email contact? Regards Bilal -- _ -- 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
Re: [asterisk-users] callerid overwrite
On 1/28/14, 1:55 PM, motty cruz wrote:
Hi all,
I'm having issues with overwrite caller id, when I call someone my
caller id should be mycompanyinc but instead my id shows up as my
extension number 101.
this is what i have in sip.conf
[101]
type=friend
context=sipphones
call-limit=99
callerid=iuser 101
disallow=all
allow=ulaw
allow=alaw
username=101
secret=Passwd
dtmfmode=rfc2833
host=dynamic
mailbox=101@default
nat=yes
canreinvite=no
this is what i have in extensions.conf
[outbound]
exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)
This is how we have it and it works fine on Asterisk 1.8:
Set(CALLERID(number)=insert your number here)
exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)
any ideas? as this happens random,
--
Technical Support
http://www.cellroute.net
--
_
-- 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] Integration with outlook
We have used this commercial software to dial via our IP phones at my office. It's about $10 a license IIRC. http://www.theteletrigger.com/ On Tue, Jan 28, 2014 at 12:13 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; Is there a method way to be able to dial the phone number through asterisk from the outlook email contact? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callerid overwrite
On Tue, 28 Jan 2014 10:55:58 -0800
motty cruz motty.c...@gmail.com wrote:
this is what i have in extensions.conf
[outbound]
exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)
exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)
any ideas? as this happens random,
You're setting CALLERID(num) to a name. Use CALLERID(name) instead.
Additionally, you might want to set CALLERID(num) to your DID.
You can do both name and number at the same time by using
CALLERID(all), something like this:
exten = _91NXXNXX,1,Set(CALLERID(all)=mycompanyinc123-456-7890)
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callerid overwrite
Thank you for your reply, I updated extensions.conf file to reflect your
suggestion, I will monitor Asterisk for any more issues,
Thanks,
On Tue, Jan 28, 2014 at 11:23 AM, Andres and...@telesip.net wrote:
On 1/28/14, 1:55 PM, motty cruz wrote:
Hi all,
I'm having issues with overwrite caller id, when I call someone my caller
id should be mycompanyinc but instead my id shows up as my extension
number 101.
this is what i have in sip.conf
[101]
type=friend
context=sipphones
call-limit=99
callerid=iuser 101
disallow=all
allow=ulaw
allow=alaw
username=101
secret=Passwd
dtmfmode=rfc2833
host=dynamic
mailbox=101@default
nat=yes
canreinvite=no
this is what i have in extensions.conf
[outbound]
exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)
This is how we have it and it works fine on Asterisk 1.8:
Set(CALLERID(number)=insert your number here)
exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)
any ideas? as this happens random,
--
Technical Supporthttp://www.cellroute.net
--
_
-- 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] Message udevd[374]: timeout: killing '/sbin/modprobe -b dans syslog
Bonjour, Depuis que j'ai passé une machine de Squeeze à Wheezy, j'ai toutes les secondes dans /var/log/syslog, ce message : udevd[374]: timeout: killing '/sbin/modprobe -b x86cpu:vendor:0002:family:0014:model:0002:feature:,,0001,0002,0003,0004,0005,0006,0007,0008,0009,000B,000C,000D,000E,000F,0010,0011,0013,0017,0018,0019,001A,001C,0020,0021,0022,0023,0024,0025,0026,0027,0028,0029,002B,002C,002D,002E,002F,0030,0031,0034,0036,0037,0038,0039,003A,003B,003D,0064,0068,006A,0071,0078,007A,007C,0080,0083,0089,008D,0097,00C0,00C1,00C2,00C3,00C4,00C5,00C6,00C7,00C8,00CA,00CC,00CD,00E1,00E8,0105,0106,0107,0108,010D#012' [447] Que signifie-t-il ? Qu'est ce x86cpu:vendor:0002:family:0014:model:0002:feature ? Une idée ? Slts -- _ -- 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] Integration with outlook
There's also SIPTAPI: http://www.ipcom.at/en/telephony/siptapi/ On Tue, Jan 28, 2014 at 7:13 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; Is there a method way to be able to dial the phone number through asterisk from the outlook email contact? Regards Bilal -- _ -- 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] Sorry [Was Re: Message udevd[374]: timeout: killing '/sbin/modprobe -b dans syslog]
Wrong list, wrong language : double appologize for my previous message .. 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
[asterisk-users] Adding Berkeley DB to Asterisk 1.8 and above
All; I'm working on a project (using Asterisk 1.8, but 11 would probably work just as well) where so far I've been able to originate over 1,000 concurrent outbound faxes. I have no problem with that so far. Where I have the problem is that Asterisk is dumping core after the faxes are sent. Now two things happen after the faxes are sent. (1) A fax log similar to a CDR is written to MySQL for each fax transmitted. Think of it as a fax-centric CDR that only contains information related to the fax just sent. (2) The Asterisk CDR is written to MySQL. The problem appears to be with the CDR's because if I do not record them, things seem to appear to work correctly. I'm going to try a remote MySQL server on a second machine and that may or may not work, but I don't want to have to come up with the expense of a second server if I could avoid it. I tried using Sqlite for the CDR's but the concurrency issues were a killer. What I would really like to do is add the Berkeley DB to Asterisk. I'm thinking that by using the Berkeley DB, I could most likely write the fax logs and CDR's without having the system even breathing heavy. The problem is that I have no idea how to add the BDB to Asterisk. Can someone point me in the right direction as far as documentation and examples go? I would greatly appreciate it and will make it all available publically if the implementation turns out well. Thanks; John Tech Support Tech Support VoIP Business Solutions 240-215-3479 (Work/Fax) supp...@voipbusiness.us mailto:f...@voipbusiness.us -- _ -- 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
