[asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
Hi, I've seen that function CURL is missing from 1.8 but back in with 10 (see wiki.asterisk.org). With asterisk 1.8 and above, for a custom CID Name lookup application, which is the most efficient way to send an HTTP GET from the dialplan and parse its response (code and content) ? 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] Q: SIPNATtraversal.pdf
Hello, To understand SIP and NAT Traversal better I'm looking for a PDF doc with the name SIPNATtraversal.pdf; one can find many pointers and recomendations like this: http://www.sipcenter.com/sip.nsf/html/WEBB5YN5GE/$FILE/SIPNATtraversal.pdf. but the URL is outdated; anybody here with a working pointer or who could mail me a copy? Thanks matthias -- Matthias Apitz e g...@unixarea.de - w http://www.unixarea.de/ UNIX since V7 on PDP-11, UNIX on mainframe since ESER 1055 (IBM /370) UNIX on x86 since SVR4.2 UnixWare 2.1.2, FreeBSD since 2.2.5 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP and NAT best practices since recent changes?
Hi, Since the recent update to the NAT configuration options and defaults in chan_sip.so, I am interested in any SIP/NAT best practices advice. What I've always done in the past is: Global: nat=no SIP handsets that are local: nat=no SIP handsets that are remote: nat=yes ITSP SIP trunks: nat=yes I will then set externip and localnet to reflect the local setup, UNLESS there is a functional SIP ALG doing the work in the gateway device. I make this statement because I've found one or two firewalls where it is best to disable the SIP ALG, and one or two where it is best to leave it enabled. The above always worked very well, but I now find my asterisk logs being spammed with warnings containing lots of !! and I'd like to know the best way to operate to achieve what I've always had while following the new rules in order to be as secure as possible with clean logs. I should add that we do not accept unsolicited connections, and 99% of attempts to connect will be stopped at the firewall. Thanks, Steve -- _ -- 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] Iax hold events in AMI 1.1
Hello all, In my PBX, AMI does not generate Hold or MoH events for a IAX trunk. I did the following tests with different channels: Dahdi to SIP: Hold and MoH events. Dahdi to Dahdi: MoH events. SIP to SIP:Hold and MoH events. IAX to SIP:Hold events. IAX to Dahdi: No events. For me, It seems odd that Dadhdi and SIP generate events and IAX doesn't. Am I doing something wrong in manager or iax configuration files? Thanks in advance, Alex Asterisk version: 1.8.7.1 Operation system: CentOS release 5.5 manager.conf: [general] displaysystemname = yes enabled = yes webenabled = yes port = 5038 httptimeout = 60 bindaddr = 0.0.0.0 [alex] secret = alex read = system,call,log,verbose,command,agent,user,config write =system,call,log,verbose,command,agent,user,config,originate Sip.conf: [general] callerid=PBX_Athens context=default allowoverlap=no realm=athens.lab.colours.local bindport=5060 bindaddr=0.0.0.0 srvlookup=yes notifyringing=yes notifyhold=yes notifycid=yes callevents=yes limitonpeers=yes rtcachefriends=yes allowsubscribe=yes subscribecontext=internal_hints call-limit=2 [1001] type=friend callerid=1001 1001 context=internal host=dynamic disallow=all allow=alaw qualify=yes callgroup=1 pickupgroup=1 call-limit=2 Iax.conf: [iax_trunk] type=friend context=from_iax disallow=all allow=alaw qualify=yes host=rome.lab.colours.local chan_dahdi.conf: [trunkgroups] [channels] usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no relaxdtmf=yes language=pt group= callgroup=1 pickupgroup=1 threewaycalling=yes transfer=yes signalling=fxo_ks callerid=Zap 1004 context=internal channel = 4 -- _ -- 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] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
Olivier oza_4...@yahoo.fr writes: I've seen that function CURL is missing from 1.8 but back in with 10 (see wiki.asterisk.org). I see the CURL function in Asterisk 1.8.7.1, found in the res_curl module. In Fedora it is available in a separate package called asterisk-curl. If you do not get res_curl, it is likely because a prerequisite library was not installed when you built Asterisk. Try looking at MENUSELECT_DEPSFAILED in menuselect.makeopts. /Benny -- _ -- 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 audio available on SIP/172.16.129.13:5060-00000001??
Hello, Actually I have changed asterisk in such a way that any call that comes onto asterisk server will go into the voicemail() application for that user. I am sending the media through SIPp by putting the following action in scenario file: !-- Play a pre-recorded PCAP file (RTP stream) -- nop action exec play_pcap_audio=pcap/g711a.pcap/ /action /nop Regards, Shalu Date: Wed, 11 Jan 2012 10:59:33 +0530 From: virendra bhati virbh...@gmail.com Subject: Re: [asterisk-users] No audio available on SIP/172.16.129.13:5060-0001?? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: cannhuhdoqvvovvyib7s0pnj+or_xy94d0yl8fe71eka+f4d...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi Shalu, How you are invoking call in dialplan. it's completely depends on that. And error show that no voice is there for store in voicemail . On Wed, Jan 11, 2012 at 10:05 AM, shalu dhamija shalu.dham...@rancoretech.com wrote: Hello, I am trying to run load on asterisk server(version 1.8.7.1) for the voicemail() application using SIPp tool. I am just running sipp at call rate of 1 cps with the following command: ./sipp -m 9000 -r 1 -rp 1000 -d 45 -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 1 172.16.129.14 --trace_err I am trying to deposit 9000 messages in the mailbox of user 1 (given by the -s option) but the following warning is coming on the asterisk server due to which the message does not get deposited into the users mailbox: No audio available on SIP/172.16.129.13:5060-0001?? I have set rtpstart=6000 and rtpend=2 in rtp.conf. Can someone please let me know how to avoid these kind of warnings. Thanks. Shalu -- _ -- 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] Anyone have a reliable T.38 Solution
On 01/11/2012 03:01 PM, Olivier wrote: 2012/1/5, Kevin P. Flemingkpflem...@digium.com: On 01/04/2012 12:25 AM, Matt Darnell wrote: Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI-- Asterisk-- T.38-- ATA-- Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! What you are looking for is T.38 gateway mode (converting between T.30 over modems on a TDM circuit and T.38 over UDPTL), and the answer is no: Asterisk 1.8 does not have T.38 gateway mode. Asterisk 10 does, and it is supported using SpanDSP and res_fax_spandsp. It is not yet supported by Digium's Fax for Asterisk commercial FAX module. Do you have any idea when Digium's Fax for Asterisk commercial FAX module could roughly become supported ? Are you really desperate to pay for functionality you can get for free? Steve -- _ -- 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] OT - Which iceweasel plugin to play gsm sound files ?
Hi, Which plugin can I add to my iceweasel browser (debian squeeze) to play gsm sound files ? Cheers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Q: SIPNATtraversal.pdf
What about this http://support.avaya.com/css/P8/documents/100102120 or this? http://www.ingate.com/files/Solving_Firewall-NAT_Traversal.pdf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthias Apitz Sent: Wednesday, January 11, 2012 4:05 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Q: SIPNATtraversal.pdf Hello, To understand SIP and NAT Traversal better I'm looking for a PDF doc with the name SIPNATtraversal.pdf; one can find many pointers and recomendations like this: http://www.sipcenter.com/sip.nsf/html/WEBB5YN5GE/$FILE/SIPNATtraversal.pdf. but the URL is outdated; anybody here with a working pointer or who could mail me a copy? Thanks matthias -- Matthias Apitz e g...@unixarea.de - w http://www.unixarea.de/ UNIX since V7 on PDP-11, UNIX on mainframe since ESER 1055 (IBM /370) UNIX on x86 since SVR4.2 UnixWare 2.1.2, FreeBSD since 2.2.5 -- _ -- 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] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
I think the wiki may have just missed func_curl. I have a couple 1.8.x machines with working func_curl. Have you tried to compile it anyway? Thanks!! - Original message - Hi, I've seen that function CURL is missing from 1.8 but back in with 10 (see wiki.asterisk.org). With asterisk 1.8 and above, for a custom CID Name lookup application, which is the most efficient way to send an HTTP GET from the dialplan and parse its response (code and content) ? 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 -- _ -- 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] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
Yes, you're right, I can read this menuselect.makeopts : MENUSELECT_DEPSFAILED=MENUSELECT_RES=res_curl I should investigate why res_curl was not built. 1. But, on your own 1.8.7 system, do you have something related to CURL when typing core show functions (or core show applications) ? I'm asking because func_CURL is missing from https://wiki.asterisk.org/wiki/display/AST/Dialplan+Functions (asterisk 1.8 version) which is misleading. 2. How would you rate CURL function performance ? Would you recommend it (for CID Lookup, for instance) ? 2012/1/11, Benny Amorsen benny+use...@amorsen.dk: Olivier oza_4...@yahoo.fr writes: I've seen that function CURL is missing from 1.8 but back in with 10 (see wiki.asterisk.org). I see the CURL function in Asterisk 1.8.7.1, found in the res_curl module. In Fedora it is available in a separate package called asterisk-curl. If you do not get res_curl, it is likely because a prerequisite library was not installed when you built Asterisk. Try looking at MENUSELECT_DEPSFAILED in menuselect.makeopts. /Benny -- _ -- 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] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? [SOLVED]
2012/1/11, Jim DeVito asterisk-users-mailing-l...@devito.cc: I think the wiki may have just missed func_curl. I have a couple 1.8.x machines with working func_curl. Have you tried to compile it anyway? At the time I first wrote my question, libcurl4-openssl-dev was missing from my system so func_CURL was not available, which lead me check with wiki.asterisk.org. Now, I added the missing library and I can see CURL function available. I will open a ticket to let concerned people know about the missing entry in wiki.asterisk.org Thanks you very much for your help ! Thanks!! - Original message - Hi, I've seen that function CURL is missing from 1.8 but back in with 10 (see wiki.asterisk.org). With asterisk 1.8 and above, for a custom CID Name lookup application, which is the most efficient way to send an HTTP GET from the dialplan and parse its response (code and content) ? 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 -- _ -- 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] Anyone have a reliable T.38 Solution
2012/1/11, Steve Underwood ste...@coppice.org: On 01/11/2012 03:01 PM, Olivier wrote: 2012/1/5, Kevin P. Flemingkpflem...@digium.com: On 01/04/2012 12:25 AM, Matt Darnell wrote: Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI-- Asterisk-- T.38-- ATA-- Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! What you are looking for is T.38 gateway mode (converting between T.30 over modems on a TDM circuit and T.38 over UDPTL), and the answer is no: Asterisk 1.8 does not have T.38 gateway mode. Asterisk 10 does, and it is supported using SpanDSP and res_fax_spandsp. It is not yet supported by Digium's Fax for Asterisk commercial FAX module. Do you have any idea when Digium's Fax for Asterisk commercial FAX module could roughly become supported ? Are you really desperate to pay for functionality you can get for free? Not yet ;-))) But the increased fax sending speed (14.4 kbs/s says the datasheet but I must be too naive to still read datasheets) may be a feature interesting for some. By the way, which spandsp version would recommend for asterisk 10 ? spandsp-0.0.6pre18.tgz ? Steve -- _ -- 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] Anyone have a reliable T.38 Solution
On 01/11/2012 09:16 AM, Olivier wrote: 2012/1/11, Steve Underwoodste...@coppice.org: On 01/11/2012 03:01 PM, Olivier wrote: 2012/1/5, Kevin P. Flemingkpflem...@digium.com: On 01/04/2012 12:25 AM, Matt Darnell wrote: Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI--Asterisk--T.38--ATA--Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! What you are looking for is T.38 gateway mode (converting between T.30 over modems on a TDM circuit and T.38 over UDPTL), and the answer is no: Asterisk 1.8 does not have T.38 gateway mode. Asterisk 10 does, and it is supported using SpanDSP and res_fax_spandsp. It is not yet supported by Digium's Fax for Asterisk commercial FAX module. Do you have any idea when Digium's Fax for Asterisk commercial FAX module could roughly become supported ? Are you really desperate to pay for functionality you can get for free? Not yet ;-))) But the increased fax sending speed (14.4 kbs/s says the datasheet but I must be too naive to still read datasheets) may be a feature interesting for some. By the way, which spandsp version would recommend for asterisk 10 ? spandsp-0.0.6pre18.tgz ? There are currently no modem speed differences between res_fax_spandsp and res_fax_digium. Both support all commonly-used FAX modems except V.34. We do not currently have an estimate on when Fax For Asterisk will support T.38 gateway mode (or V.34, for that matter). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP and NAT best practices since recent changes?
On 01/11/2012 05:29 AM, Steve Davies wrote: Hi, Since the recent update to the NAT configuration options and defaults in chan_sip.so, I am interested in any SIP/NAT best practices advice. What I've always done in the past is: Global: nat=no SIP handsets that are local: nat=no SIP handsets that are remote: nat=yes ITSP SIP trunks: nat=yes I will then set externip and localnet to reflect the local setup, UNLESS there is a functional SIP ALG doing the work in the gateway device. I make this statement because I've found one or two firewalls where it is best to disable the SIP ALG, and one or two where it is best to leave it enabled. The above always worked very well, but I now find my asterisk logs being spammed with warnings containing lots of !! and I'd like to know the best way to operate to achieve what I've always had while following the new rules in order to be as secure as possible with clean logs. I should add that we do not accept unsolicited connections, and 99% of attempts to connect will be stopped at the firewall. The simplest answer is to always use 'nat=yes' (or at least 'nat=force_rport' in recent versions of Asterisk that support it), until you come across a SIP endpoint that fails to work properly with that setting. If you do come across such an endpoint, try hard to get it to work with that setting; if you can't, then set 'nat=no' for that endpoint, and understand that the endpoint's name could be discoverable using the attack methods previously disclosed. If the endpoint's configuration is suitably locked down (permit/deny, for example) this may not be a concern for you. If it's not locked down (for example, if it has to register to your Asterisk server from random locations), then the next step would be to seriously consider requesting that the user of that endpoint consider switching to some other SIP endpoint. To date, the only endpoints that have been identified that do *not* work with Asterisk's 'rport' handling forced upon them are Cisco phones. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone have a reliable T.38 Solution
On 01/11/2012 11:16 PM, Olivier wrote: 2012/1/11, Steve Underwoodste...@coppice.org: On 01/11/2012 03:01 PM, Olivier wrote: 2012/1/5, Kevin P. Flemingkpflem...@digium.com: On 01/04/2012 12:25 AM, Matt Darnell wrote: Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI--Asterisk--T.38--ATA--Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! What you are looking for is T.38 gateway mode (converting between T.30 over modems on a TDM circuit and T.38 over UDPTL), and the answer is no: Asterisk 1.8 does not have T.38 gateway mode. Asterisk 10 does, and it is supported using SpanDSP and res_fax_spandsp. It is not yet supported by Digium's Fax for Asterisk commercial FAX module. Do you have any idea when Digium's Fax for Asterisk commercial FAX module could roughly become supported ? Are you really desperate to pay for functionality you can get for free? Not yet ;-))) But the increased fax sending speed (14.4 kbs/s says the datasheet but I must be too naive to still read datasheets) may be a feature interesting for some. By the way, which spandsp version would recommend for asterisk 10 ? spandsp-0.0.6pre18.tgz ? How is 14.4k an increase? Both spandsp and the Digium modules do 14.4k. There is nothing the Digium module does which spandsp does not do, and the file handling in spandsp is more flexible. spandsp-0.0.6pre18.tgz is currently the right version to use? Steve -- _ -- 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] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? [SOLVED]
On Wednesday 11 January 2012, Olivier wrote: At the time I first wrote my question, libcurl4-openssl-dev was missing from my system so func_CURL was not available, which lead me check with wiki.asterisk.org. It's *always* a -dev (or -devel if you're into RPMs) package missing. Always! Frankly, why distributions still insist to separate out development files in 2012 is a mystery to me. Ubuntu especially have *no* excuse; user- friendliness is supposed to be their USP, and compiling a package from Source Code is something everyone has to do at some stage. Now, I added the missing library and I can see CURL function available. I will open a ticket to let concerned people know about the missing entry in wiki.asterisk.org Isn't the point of a Wiki, so that anybody can edit it without raising a support ticket? -- AJS Answers come *after* questions. -- _ -- 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] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? [SOLVED]
On 01/11/2012 11:14 AM, A J Stiles wrote: On Wednesday 11 January 2012, Olivier wrote: At the time I first wrote my question, libcurl4-openssl-dev was missing from my system so func_CURL was not available, which lead me check with wiki.asterisk.org. It's *always* a -dev (or -devel if you're into RPMs) package missing. Always! Frankly, why distributions still insist to separate out development files in 2012 is a mystery to me. Ubuntu especially have *no* excuse; user- friendliness is supposed to be their USP, and compiling a package from Source Code is something everyone has to do at some stage. I suspect that 99% of Ubuntu users have never built a package from source, and wouldn't even have a clue how to begin to do so. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP and NAT best practices since recent changes?
On 11 January 2012 15:43, Kevin P. Fleming kpflem...@digium.com wrote: On 01/11/2012 05:29 AM, Steve Davies wrote: Hi, Since the recent update to the NAT configuration options and defaults in chan_sip.so, I am interested in any SIP/NAT best practices advice. What I've always done in the past is: Global: nat=no SIP handsets that are local: nat=no SIP handsets that are remote: nat=yes ITSP SIP trunks: nat=yes I will then set externip and localnet to reflect the local setup, UNLESS there is a functional SIP ALG doing the work in the gateway device. I make this statement because I've found one or two firewalls where it is best to disable the SIP ALG, and one or two where it is best to leave it enabled. The above always worked very well, but I now find my asterisk logs being spammed with warnings containing lots of !! and I'd like to know the best way to operate to achieve what I've always had while following the new rules in order to be as secure as possible with clean logs. I should add that we do not accept unsolicited connections, and 99% of attempts to connect will be stopped at the firewall. The simplest answer is to always use 'nat=yes' (or at least 'nat=force_rport' in recent versions of Asterisk that support it), until you come across a SIP endpoint that fails to work properly with that setting. If you do come across such an endpoint, try hard to get it to work with that setting; if you can't, then set 'nat=no' for that endpoint, and understand that the endpoint's name could be discoverable using the attack methods previously disclosed. If the endpoint's configuration is suitably locked down (permit/deny, for example) this may not be a concern for you. If it's not locked down (for example, if it has to register to your Asterisk server from random locations), then the next step would be to seriously consider requesting that the user of that endpoint consider switching to some other SIP endpoint. To date, the only endpoints that have been identified that do *not* work with Asterisk's 'rport' handling forced upon them are Cisco phones. Excellent. Thanks as always Kevin. (Why am I not surprised about Cisco!) Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP and NAT best practices since recent changes?
From: Steve Davies davies...@gmail.com Sent: Wednesday, January 11, 2012 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP and NAT best practices since recent changes? On 11 January 2012 15:43, Kevin P. Fleming kpflem...@digium.com wrote: On 01/11/2012 05:29 AM, Steve Davies wrote: Hi, Since the recent update to the NAT configuration options and defaults in chan_sip.so, I am interested in any SIP/NAT best practices advice. What I've always done in the past is: Global: nat=no SIP handsets that are local: nat=no SIP handsets that are remote: nat=yes ITSP SIP trunks: nat=yes I will then set externip and localnet to reflect the local setup, UNLESS there is a functional SIP ALG doing the work in the gateway device. I make this statement because I've found one or two firewalls where it is best to disable the SIP ALG, and one or two where it is best to leave it enabled. The above always worked very well, but I now find my asterisk logs being spammed with warnings containing lots of !! and I'd like to know the best way to operate to achieve what I've always had while following the new rules in order to be as secure as possible with clean logs. I should add that we do not accept unsolicited connections, and 99% of attempts to connect will be stopped at the firewall. The simplest answer is to always use 'nat=yes' (or at least 'nat=force_rport' in recent versions of Asterisk that support it), until you come across a SIP endpoint that fails to work properly with that setting. If you do come across such an endpoint, try hard to get it to work with that setting; if you can't, then set 'nat=no' for that endpoint, and understand that the endpoint's name could be discoverable using the attack methods previously disclosed. If the endpoint's configuration is suitably locked down (permit/deny, for example) this may not be a concern for you. If it's not locked down (for example, if it has to register to your Asterisk server from random locations), then the next step would be to seriously consider requesting that the user of that endpoint consider switching to some other SIP endpoint. To date, the only endpoints that have been identified that do *not* work with Asterisk's 'rport' handling forced upon them are Cisco phones. Excellent. Thanks as always Kevin. (Why am I not surprised about Cisco!) Regards, Steve Steve I can't get my grandstream phones to work with force_rport behind a pfsense firewall. but yes and comedia work fine. Bryant -- _ -- 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] Exceptionally long voice queue length
I'm seeing this error thousands of times per minute and it's causing the CPU to sky rocket WARNING[16095]: channel.c:1039 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/*7...etc... Any idea what could be causing this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP and NAT best practices since recent changes?
On 01/11/2012 12:09 PM, Bryant Zimmerman wrote: *From*: Steve Davies davies...@gmail.com *Sent*: Wednesday, January 11, 2012 12:51 PM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] SIP and NAT best practices since recent changes? On 11 January 2012 15:43, Kevin P. Fleming kpflem...@digium.com wrote: On 01/11/2012 05:29 AM, Steve Davies wrote: Hi, Since the recent update to the NAT configuration options and defaults in chan_sip.so, I am interested in any SIP/NAT best practices advice. What I've always done in the past is: Global: nat=no SIP handsets that are local: nat=no SIP handsets that are remote: nat=yes ITSP SIP trunks: nat=yes I will then set externip and localnet to reflect the local setup, UNLESS there is a functional SIP ALG doing the work in the gateway device. I make this statement because I've found one or two firewalls where it is best to disable the SIP ALG, and one or two where it is best to leave it enabled. The above always worked very well, but I now find my asterisk logs being spammed with warnings containing lots of !! and I'd like to know the best way to operate to achieve what I've always had while following the new rules in order to be as secure as possible with clean logs. I should add that we do not accept unsolicited connections, and 99% of attempts to connect will be stopped at the firewall. The simplest answer is to always use 'nat=yes' (or at least 'nat=force_rport' in recent versions of Asterisk that support it), until you come across a SIP endpoint that fails to work properly with that setting. If you do come across such an endpoint, try hard to get it to work with that setting; if you can't, then set 'nat=no' for that endpoint, and understand that the endpoint's name could be discoverable using the attack methods previously disclosed. If the endpoint's configuration is suitably locked down (permit/deny, for example) this may not be a concern for you. If it's not locked down (for example, if it has to register to your Asterisk server from random locations), then the next step would be to seriously consider requesting that the user of that endpoint consider switching to some other SIP endpoint. To date, the only endpoints that have been identified that do *not* work with Asterisk's 'rport' handling forced upon them are Cisco phones. Excellent. Thanks as always Kevin. (Why am I not surprised about Cisco!) Regards, Steve Steve I can't get my grandstream phones to work with force_rport behind a pfsense firewall. but yes and comedia work fine. That's rather strange, since 'yes' includes 'force_rport'. Can you describe what 'not work' means in this case? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
Olivier oza_4...@yahoo.fr writes: 1. But, on your own 1.8.7 system, do you have something related to CURL when typing core show functions (or core show applications) ? I'm asking because func_CURL is missing from https://wiki.asterisk.org/wiki/display/AST/Dialplan+Functions (asterisk 1.8 version) which is misleading. == 8 == ursa*CLI core show version Asterisk 1.8.7.1 built by mockbuild @ x86-02.phx2.fedoraproject.org on a x86_64 running Linux on 2011-10-17 21:15:10 UTC ursa*CLI core show function CURL -= Info about function 'CURL' =- [Synopsis] Retrieves the contents of a URL == 8 == The Wiki documentation is sadly not perfect yet. /Benny -- _ -- 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] Attempt to Originate between IAX2/xxxx and an application hangs until timeout in 1.8.8.1
I am investigating an issue with IAX2 extensions in Asterisk 1.8.x. My application connects to Asterisk via AMI and attempts to run an Originate command between an extension (such as SIP/ or IAX2/) and an application (in my case it is AgentLogin). This works correctly for SIP extensions, in all Asterisk versions. With IAX2 extensions, this worked correctly in Asterisk 1.6.2.20, but started failing sometimes in Asterisk 1.8.7, and now happens every time in Asterisk 1.8.8.1. I found out that any application (not just AgentLogin) will trigger the issue. Instead of successfully ringing the IAX2 extension, as expected, the Originate attempt just sits there. The IAX2 extension does not receive any ringing indication. I can reproduce the issue by running the following command from the Asterisk console: originate IAX2/1099 application playback demo-congrats This is supposed to ring the extension, and upon picking up, should play the audio file. Instead, the IAX2 extension sits idle. Also, the Asterisk console becomes unresponsive. If I try to execute any other command (such as iax2 show threads, or even help), I get a prompt back but no command output. Then, after some time (the ring timeout, maybe), I get the output of all commands I issued during the hang. When my application connects to AMI and runs the Originate command, it eventually gets a Hangup event, as if the extension never picked up the ringing. But actually the ringing never made it to the IAX2 extension. We have noticed that the IAX2 extension itself can place calls to a SIP extension normally during the Originate hang, but it cannot receive a call from another SIP extension (Busy Here). When not attempting the Originate call, the IAX2 extension appears to behave normally. This has been triggered in three machines to date: a big server with some 40 IAX2 extensions, and two test machines (one physical and one virtual machine). Before I get into a bug hunt, I would like to know: Is this a known issue? Are there any pointers on where to look first, or what to look for, based on my symptoms? Testing with Asterisk 1.8.8.1 x86_64 and Zoiper as an IAX2 client. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
Im using the one that comes with Ubuntu Server 10.10 (0.0.6~pre12-1): http://packages.ubuntu.com/search?keywords=libspandspsearchon=namessuite=mavericksection=all And having a sweet time with T.38 gateway. Oneiric already offers latest pre18. *José Pablo Méndez * On Wed, Jan 11, 2012 at 12:39 AM, Olivier oza_4...@yahoo.fr wrote: Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting differences between them. So I prefer to double check ask for recommendations. 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 -- _ -- 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] Problems faced in load testing of asterisk
I have given the rtp port range as 6000 to 8000 in rtp.conf. Is this not sufficient for running 1000 calls. Only even ports will be used for RTP I think, odd ports are reserved for RTCP, although I don't know how SIPp behaves in this line. 2000 ports should be reduced to 1000 ports following my theory. -- _ -- 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] Attempt to Originate between IAX2/xxxx and an application hangs until timeout in 1.8.8.1
El 11/01/12 15:37, Alex Villacís Lasso escribió: I am investigating an issue with IAX2 extensions in Asterisk 1.8.x. My application connects to Asterisk via AMI and attempts to run an Originate command between an extension (such as SIP/ or IAX2/) and an application (in my case it is AgentLogin). This works correctly for SIP extensions, in all Asterisk versions. With IAX2 extensions, this worked correctly in Asterisk 1.6.2.20, but started failing sometimes in Asterisk 1.8.7, and now happens every time in Asterisk 1.8.8.1. I found out that any application (not just AgentLogin) will trigger the issue. Instead of successfully ringing the IAX2 extension, as expected, the Originate attempt just sits there. The IAX2 extension does not receive any ringing indication. I can reproduce the issue by running the following command from the Asterisk console: originate IAX2/1099 application playback demo-congrats This is supposed to ring the extension, and upon picking up, should play the audio file. Instead, the IAX2 extension sits idle. Also, the Asterisk console becomes unresponsive. If I try to execute any other command (such as iax2 show threads, or even help), I get a prompt back but no command output. Then, after some time (the ring timeout, maybe), I get the output of all commands I issued during the hang. When my application connects to AMI and runs the Originate command, it eventually gets a Hangup event, as if the extension never picked up the ringing. But actually the ringing never made it to the IAX2 extension. We have noticed that the IAX2 extension itself can place calls to a SIP extension normally during the Originate hang, but it cannot receive a call from another SIP extension (Busy Here). When not attempting the Originate call, the IAX2 extension appears to behave normally. This has been triggered in three machines to date: a big server with some 40 IAX2 extensions, and two test machines (one physical and one virtual machine). Before I get into a bug hunt, I would like to know: Is this a known issue? Are there any pointers on where to look first, or what to look for, based on my symptoms? Testing with Asterisk 1.8.8.1 x86_64 and Zoiper as an IAX2 client. Some additional information - it is the Originate with IAX2 channel that has problems, not just applications. Given that 1099 is an IAX2 extension and 1065 is a SIP extension, with FreePBX contexts, i found that originate IAX2/1099 extension 1065@from-internal hangs, but originate SIP/1065 extension 1099@from-internal succeeds. -- _ -- 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] Problems with codec translation when using Monitor and MixMonitor
Hi folks, I'm having problems when I try to record my calls using MixMonitor or Monitor. Calls are working well and audio quality is good. But I just can't get recorded audio in one leg with both applications. It happens with internal calls too. As it seems, the problem is my g729 licensing escheme. (just one license installed) What is the least number of licenses that are needed per recorded call? or what can I do to fix it? My asterisk version: Asterisk 1.8.7.1 Logs with MixMonitor: [Jan 11 17:55:48] WARNING[19500]: translate.c:256 ast_translator_build_path: No translator path from alaw to unknown [Jan 11 17:55:48] WARNING[19500]: translate.c:256 ast_translator_build_path: No translator path from alaw to unknown testpbx*CLI g729 show licenses 0/1 encoders/decoders of 1 licensed channels are currently in use Licenses Found: File: G729-... -- Key: G729-... -- Host-ID: ... -- Channels: 1 (Expires: ...) (OK) Logs with Monitor: [Jan 11 17:49:49] WARNING[19491]: translate.c:256 ast_translator_build_path: No translator path from alaw to g723 [Jan 11 17:49:49] WARNING[19491]: file.c:186 ast_writestream: Unable to translate to format wav49, source format g729 testpbx*CLI g729 show licenses 0/1 encoders/decoders of 1 licensed channels are currently in use Licenses Found: File: G729-...lic -- Key: G729-... -- Host-ID: ... -- Channels: 1 (Expires: ...) (OK) I've searched for this on forums but can't find a complete answer yet. Thanks! Elder D. Arohuanca Lima - Peru -- _ -- 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] Exceptionally long voice queue length
which version of Asterisk are you using !. AFAIK this issue has been in asterisk for queue calls and I'm not sure if this has ever been resolved fully and stabilized. Not binding to Local channel only, I've seen this on SIP and IAX channels as well ! On Thu, Jan 12, 2012 at 12:56 AM, Vik Killa vipki...@gmail.com wrote: I'm seeing this error thousands of times per minute and it's causing the CPU to sky rocket WARNING[16095]: channel.c:1039 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/*7...etc... Any idea what could be causing this? -- _ -- 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
