[asterisk-users] R: Asterisk V/s FreeSwitch
From the second link Gilles suggested: Aterisk vs FreeSWITCH [...] The mailing list (FreeSWITCH) is also a very nice place. In both places (IRC and mailing list) they are very friendly and supportive, unlike the Asterisk/Digium community. [...] BAH... -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Gilles Inviato: martedì 7 febbraio 2012 16:41 A: asterisk-users@lists.digium.com Oggetto: Re: [asterisk-users] Asterisk V/s FreeSwitch On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati virbh...@gmail.com wrote: Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... Provided Asterisk, even in release 1.8 or 10, does handle much fewer concurrent calls than Freeswitch, you might find the answer in those articles: How does FreeSWITCH compare to Asterisk? www.freeswitch.org/node/117 Asterisk vs FreeSWITCH www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ Asterisk vs. FreeSWITCH www.anders.com/cms/266 Open Source VoIP: Asterisk or FreeSwitch? www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233 FreeSwitch vs Asterisk www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Realtime Templates (!)
Hello, Is it possible to assign templates defined in sip.conf to sip realtime peers? There was another mail in 2008 which asked the same question but never received a response. Thanks, Alex -- _ -- 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] R: Stun Server
In debian do 'apt-get install stun'. Change /etc/default/stun. Done Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Bryant Zimmerman Inviato: mercoledì 27 luglio 2011 15:30 A: asterisk-users@lists.digium.com Oggetto: Re: [asterisk-users] Stun Server We have been running a windows stun server for 5 years now and I would like to change to either a linux of freebsd based unit to phase out the old XP box in our datacenter. What should I look at that would be a good replacement. The windows box has worked but the hardware is at end of life and I want to move it to a vm and I don't want Windows. Any advise is apperciated. Thanks zktech -- _ -- 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] Browser based SIP UA
Hello, can anyone recommend a browser based SIP client that works well with Asterisk? I need something that requires authentication (based on Asterisks peer name and pass). Thanks in advance! Alex -- _ -- 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] R: Browser based SIP UA
I mean anything not an extension that can run on Linux (Apache/Tomcat). Thanks, Alex -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Alex Balashov Inviato: martedì 26 luglio 2011 16:15 A: asterisk-users@lists.digium.com Oggetto: Re: [asterisk-users] Browser based SIP UA On 07/26/2011 10:13 AM, Alexandru Oniciuc wrote: can anyone recommend a browser based SIP client that works well with Asterisk? I need something that requires authentication (based on Asterisks peer name and pass). What do you mean browser-based? Any particular preference of technology? Flash? Silverlight? Java applet? Browser extension? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] R: How to remove asterisk ?
What do you mean? Did you installed from sources or distro packet? sources: make uninstall distro: Every distro has its own commands (yum, apt-get ecc) Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di virendra bhati Inviato: venerdì 10 giugno 2011 11:26 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: [asterisk-users] How to remove asterisk ? Hi List, Is there any way by which we can remove asterisk from machine without deleting folder manually? I did google and gets various solution by no success. even after deleted asterisk will be there . - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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] R: Nat=yes
Thank you all for your answers, I will stick with nat=yes. Alex -- _ -- 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] R: R: No Internet, no asterisk
srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet ; Specifying a port in a SIP peer definition or ; when dialing outbound calls will supress SRV ; lookups for that peer or call. Make sure you don't have ANY reference to domain names in your sip.conf, only IPs, and eventually try to specify the port as described above. I didn't tried this myself but I think this should be the way to do it (srvlookup=no). Regards, ALEX -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Niccolò Belli Inviato: martedì 19 aprile 2011 12:05 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] R: No Internet, no asterisk Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto: Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and internet is offline. srvlookup = no didn't help. What about putting my provider's name in /etc/hosts? Should it solve the problem? A caching nameserver is not a viable solution because I want it working even after a month without internet access. Cheers, Darkbasic P.S. Why nobody ever fixed this annoying bug? Is there a special reason behind? -- _ -- 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] R: No Internet, no asterisk
Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and internet is offline. Regards, Alex -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Niccolò Belli Inviato: lunedì 18 aprile 2011 12:17 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: [asterisk-users] No Internet, no asterisk Hi, this is an old outstanding problem, unfortunately I don't remember how to walkaround it. I use asterisk 1.8.3 and I have a public IP in my network interface. As soon as the Internet connection goes down, phones stop working. I want to be able to use pstn, isdn and the gsm gateway even if the Internet connection goes down, how can I achieve it? Thank you, Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Registrations
Hello list, I need some light regarding the way asterisk is handling the SIP Registration method: I have an asterisk 1.6.0.22 and a UAC that sends REGISTER requests without the Authentication part in the sip message. The UAC expects a 401 reply to create the correct auth request. When it receives an empty REGISTER asterisk does basically 2 things: 1. returns an 503 --- SIP read from UDP://1.2.3.5:5060 --- REGISTER sip:1.2.3.4;user=phone SIP/2.0 Via:SIP/2.0/UDP 1.2.3.5:5060;rport;branch=z9hG4bK62F7FE99 From:sip:123456...@1.2.3.4;tag=3D090004 To:sip:123456...@1.2.3.4 Call-ID:3D0807BC@ CSeq:53246 REGISTER Max-Forwards:70 Contact:sip:123456...@1.2.3.5:5060 Allow:INVITE,CANCEL,ACK,REGISTER,INFO,BYE,OPTIONS,REFER,NOTIFY,SUBSCRIBE,MESSAGE Expires:60 User-Agent:OTHER UA Content-Length:0 - --- (12 headers 0 lines) --- --- Transmitting (no NAT) to 1.2.3.5:5060 --- SIP/2.0 503 Server error Via: SIP/2.0/UDP 1.2.3.5:5060;rport;branch=z9hG4bK62F7FE99;received=1.2.3.5 From: sip:123456...@1.2.3.4;tag=3D090004 To: sip:123456...@1.2.3.4;tag=as7bb7cbf6 Call-ID: 3D0807BC@ CSeq: 53246 REGISTER User-Agent: Asterisk 1.6.0.22 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 2. returns a 401 with the nonce --- SIP read from UDP://1.2.3.5:5060 --- REGISTER sip:1.2.3.4;user=phone SIP/2.0 Via:SIP/2.0/UDP 1.2.3.5:5060;rport;branch=z9hG4bK62F7FEA5 From:sip:123456...@1.2.3.4;tag=3D090008 To:sip:123456...@1.2.3.4 Call-ID:3D0807BC@ CSeq:53248 REGISTER Max-Forwards:70 Contact:sip:123456...@1.2.3.5:5060 Allow:INVITE,CANCEL,ACK,REGISTER,INFO,BYE,OPTIONS,REFER,NOTIFY,SUBSCRIBE,MESSAGE Expires:60 User-Agent:OTHER UA Content-Length:0 - --- (12 headers 0 lines) --- Sending to 1.2.3.5 : 5060 (no NAT) --- Transmitting (no NAT) to 1.2.3.5:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 1.2.3.5:5060;rport;branch=z9hG4bK62F7FEA5;received=1.2.3.5 From: sip:123456...@1.2.3.4;tag=3D090008 To: sip:123456...@1.2.3.4;tag=as0811b2ee Call-ID: 3D0807BC@ CSeq: 53248 REGISTER User-Agent: Asterisk 1.6.0.22 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=myrealm, nonce=6a08a09c Content-Length: 0 The questions are: * is asterisk supposed to return the 401 to a REGISTER method which lacks the Auth Info? I saw that it returns 401 to REGISTER methods that have the wrong nonce and this behavior should be correct. * the Register method should always contain the last nonce and the auth part? * why the 503 message? Thanks in advance, Alex -- _ -- 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] R: asterisk compatible cards?
Hi Daniel, have a look at this page, maybe it will help you find a reseller: http://www.voip-info.org/wiki/view/Asterisk+Consultants+Romania . Best Regards, Alex -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Daniel Petre Inviato: lunedì 2 agosto 2010 15:36 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: [asterisk-users] asterisk compatible cards? hello, i just subscribed to this list, i discovered asterisk and i would like to try it at home on my personal pc. the computer is a p4 at 3 ghz with 2 gb ram and 80 gb hdd, a 1 Mbit guarranted connection and runs a gentoo linux. i search about digium products but i can't find them in my area on any shops, i was wondering if good people here could recommend some PCI or PCIex cards for a beginner to play with one telefonic line (which i will install it soon via provider) thanks! -- _ -- 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] Group call limit
Hello list, is it possible to group some peers and limit their overall call limit? Ex: 4 peers can make max 2 concurrent calls. Thanks in advance, Alex -- _ -- 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] Little t38 bug?
Hello List,
I think I've discovered a little bug in t.38 bug in 1.6.0.22
regarding the speed (T38MaxBitRate) used to send the faxes.
Asterisk always responds with a=T38MaxBitRate:2400. I've tried
with Patton and Grandstream devices and the result is always the same.
Patton ignores the parameter and sends the fax at 9600.
Grandstream doesn't, and all the faxes are going in and out at
2400.
Looking at the code I found this in chan_sip.c (line 7736):
if ((sscanf(a, T38FaxMaxBuffer:%30u, x) == 1)) {
ast_debug(3, MaxBufferSize:%d\n, x);
found = TRUE;
} else if ((sscanf(a, T38MaxBitRate:%30u, x) == 1) || (sscanf(a,
T38FaxMaxRate:%30u, x) == 1)) {
ast_debug(3, T38MaxBitRate: %d\n, x);
switch (x) {
case 14400:
p-t38.their_parms.rate = AST_T38_RATE_14400;
break;
case 12000:
p-t38.their_parms.rate = AST_T38_RATE_12000;
break;
case 9600:
p-t38.their_parms.rate = AST_T38_RATE_9600;
break;
case 7200:
p-t38.their_parms.rate = AST_T38_RATE_7200;
break;
case 4800:
p-t38.their_parms.rate = AST_T38_RATE_4800;
break;
case 2400:
p-t38.their_parms.rate = AST_T38_RATE_2400;
break;
}
found = TRUE;
else if {...
If I'm not misteaking the second if else condition will never be true if the
other device sends T38FaxMaxBuffer (wich they all usually do).
Shouldn't it be
if((sscanf(a, T38FaxMaxBuffer:%30u, x) == 1) ((sscanf(a,
T38MaxBitRate:%30u, x) == 0) || (sscanf(a, T38FaxMaxRate:%30u, x) == 0)))
??
Thanks,
Alex
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[asterisk-users] R: new way of asterisk and kamailio (openser) realtime integration
kb.asipto.com isn't reachable: DNS doesn't resolve the domain name. Alex -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Daniel-Constantin Mierla Inviato: lunedì 17 maggio 2010 12:01 A: asterisk-users@lists.digium.com Oggetto: [asterisk-users] new way of asterisk and kamailio (openser) realtime integration Hello, I put together a new tutorial about asterisk realtime integration with kamailio (openser). This time the database used is the one of asterisk, also call routing logic is controlled by asterisk, here is the link: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Practically is an easier way to scale starting from existing asterisk installations. The other (old) version I wrote for long time, using kamailio database and asterisk just for media services, is available at: http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x Hope is useful for some of you! Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Dialplan Failover Solution
Hello list,
I need a hand to find the best dialplan failover solution when
using two SIP Trunks.
My reasons to do failover are:
a) one of the two providers could be unreachable
b) both providers may be UP but one of them could return a SIP error
message (maybe caused by DOWN E1s)
Googling I found a few possible solutions:
1. Using DIALSTATUS variable.
2. Dialing in sequence:
exten = _X.,1,Dial(SIP/${TRUNK1}/${EXTEN})
exten = _X.,2,Dial(SIP/${TRUNK2}/${EXTEN})
3. ChanIsAvail
Using the first method it's possible to get the CONGESTION and
CHANUNAVAIL status which pretty much solves my problem but it takes more than 2
lines of dialplan(I like one liners).
The second solution requires less space in the dialplan but it should work only
when the called party is busy (or maybe even when the first trunk is down).
Is there a clean way to send the call to the second SIP provider if the first
one is unreachable or spits out sip error messages?
Thanks in advance,
Alex
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[asterisk-users] Strange initial RING
Hello list! I'm having a strange problem with the VoIP Gateway that I'm using to go on the PSTN: if the number on the other end is busy or unavailable I hear an initial RING, generated by Asterisk from what I'm seeing and after that the line goes down with busy signal: Here is the scenario: Softphone *ASTERISK PATTON PSTN [BUSY CALLED EXTENSION] 1. INVITE INVITE INVITE 2. SIP/2.0 100 Trying 3.RING SIP/2.0 180 Ringing SIP/2.0 183 Session Progress 4. SIP/2.0 603 Declined SIP/2.0 406 Not Acceptable Is this regular? Asterisk isn't supposed to generate the RING only after the first one received from the PATTON? Asterisk version: 1.6.0.22 Thank you in advance for the support. Best Regards, Alex -- _ -- 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] R: Strange initial RING
No Gordon, the 'r' parameter isn't enabled:
Dial(${TRUNK}/${EXTEN},60)
Thanks,
Alex
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Gordon Henderson
Inviato: venerdì 19 marzo 2010 10:55
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] Strange initial RING
On Fri, 19 Mar 2010, Alexandru Oniciuc wrote:
Hello list!
I'm having a strange problem with the VoIP Gateway that
I'm using to go on the PSTN: if the number on the other end is busy or
unavailable I hear an initial RING, generated by Asterisk from what I'm
seeing and after that the line goes down with busy signal:
Do you have the 'r' parameter in your Dial() instruction?
Gordon
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[asterisk-users] R: Asterisk hangup all incoming calls after 10 seconds
NOTICE[14413] rtp.c: Unknown RTP codec 126 received from '192.168.20.113' Maybe the codec 126 is the problem? [core] show codecs [core] show translation Maybe you don't have the codec required by your provider. Regards, Alex -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Giorgio Incantalupo Inviato: mercoledì 17 marzo 2010 11:04 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] Asterisk hangup all incoming calls after 10 seconds Hi Bruno, I remember one of our customer had a similar problem with tellfree in Brazil. Their IT technician told me it was due to a g729 codec problem...they installed it and the problem disappeared. I never checked, I could only trust their man. Maybe it can help. Giorgio P.S.: let me know if it works, please! Bruno Camargo wrote: Hello Gentleman, I'm new to asterisk, this is my first instalation, first post... so I'd like to apologize if this question has already been made. I googled but I couldn't find nothing similar. Here's the thing. I'm migrating from ATA to Asterisk one of my client's office and I have a very simple setup. A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a totally digital setup, it means I have no analogic cards connected. I can make calls between my extension perfectly; I can make outgoing calls without any problems; Incoming calls are dropped after exatly 10 seconds; All incoming calls. The asterisk box is hooked up to the LAN switch and it runs with a private IP address. I have another Linux box performing firewall/routing roles. Outgoing and incoming calls working perfectly from the ATA (linksys pap2t) but not from asterisk, because it hangs up after 10 seconds. Some LOGS: [Mar 16 15:11:12] DEBUG[13311] acl.c: # Testing 192.168.20.113 with 192.168.20.0 [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 0: OPTIONS sip:2...@192.168.20.113:15956;rinstance=542e2865b2c6abe1 SIP/2.0 (71) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.20.249:5060;branch=z9hG4bK29f64fe1;rport (65) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 2: From: asterisk sip:aster...@192.168.20.249 mailto:sip%3aaster...@192.168.20.249;tag=as4bdc3589 (61) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 3: To: sip:2...@192.168.20.113:15956;rinstance=542e2865b2c6abe1 (61) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 4: Contact: sip:aster...@192.168.20.249 mailto:sip%3aaster...@192.168.20.249 (38) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 5: Call-ID: 7a4676c71af6501909db830431000...@192.168.20.249 mailto:7a4676c71af6501909db830431000...@192.168.20.249 (56) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 9: Date: Tue, 16 Mar 2010 18:11:12 GMT (35) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 11: Supported: replaces (19) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 12: Content-Length: 0 (17) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.20.249:5060;branch=z9hG4bK29f64fe1;rport=5060 (70) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 2: Contact: sip:192.168.20.113:15956 http://192.168.20.113:15956 (35) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 3: To: sip:2...@192.168.20.113:15956;rinstance=542e2865b2c6abe1;tag=67747e4a (74) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 4: From: asterisksip:aster...@192.168.20.249 mailto:sip%3aaster...@192.168.20.249;tag=as4bdc3589 (60) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 5: Call-ID: 7a4676c71af6501909db830431000...@192.168.20.249 mailto:7a4676c71af6501909db830431000...@192.168.20.249 (56) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 7: Accept: application/sdp (23) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 8: Accept-Language: en (19) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 10: User-Agent: X-Lite release 1104o stamp 56125 (44) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 11: Content-Length: 0 (17) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 12: (0) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: = Found Their Call ID:
[asterisk-users] R: Can not enable sip debug because CLI flooded
Edit logger.conf and set the desired log level. To disable the messages below just remove the severity notice from console. console = notice,warning,error,debug Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di jonas kellens Inviato: venerdì 12 marzo 2010 10:20 A: Asterisk Mailing Oggetto: [asterisk-users] Can not enable sip debug because CLI flooded Hello list, I have nat=no and qualify=no in my sip peer definition and still my CLI is flooded with : [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (24ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (25ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (29ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (21ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (23ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (21ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (29ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (22ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (27ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (31ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (22ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (21ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (23ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (25ms / 2000ms) [Mar 12 10:17:27] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (27ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (23ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (22ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (22ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (39ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (37ms / 2000ms) [Mar 12 10:17:28] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (37ms / 2000ms) I want to enable sip debugging but then my CLI is even more flooded with al the SIP OPTION packets... What can I do ?? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] t38 ATA
Hello, I need a hand in choosing a small ATA, even with one FXS port, that should do only fax with T38. I've tried Grandstream (ht286 model) but the faxes go out without ECM, even if the Fax machine has ECM enabled. Is there anyone that can recommend an ATA that might do the trick? Thanks, Alex -- _ -- 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] R: t38 ATA
Hi Steve, the remote device is an Hylafax Server that does ECM. The sending fax device, that's attached to the ATA, is a Philips fax machine with ECM enabled. If I send with the same machine but attached to a Patton 4114 with T38 enabled my faxes go to the other end with ECM enabled and with no errors. I've also tested the Grandstream ATA with analog fax machines attached to the PSTN and the results are the same: bad quality faxes or no faxes at all. Again, tested with the Patton device I get no errors. In this case I think it's the Grandstream. Best Regards, Alex Da: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] per conto di Steve Underwood [ste...@coppice.org] Inviato: venerdì 12 marzo 2010 18.01 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] t38 ATA On 03/13/2010 12:00 AM, Alexandru Oniciuc wrote: Hello, I need a hand in choosing a small ATA, even with one FXS port, that should do only fax with T38. I’ve tried Grandstream (ht286 model) but the faxes go out without ECM, even if the Fax machine has ECM enabled. Is there anyone that can recommend an ATA that might do the trick? Thanks, Alex The Grandstreams do T.38 quite well, and they do support ECM. It is probably your service provider which is blocking ECM. Many of them do. People complain a lot about Grandstream, but its mostly their phones. Their ATA are amongst the better ones. Sadly, that doesn't mean an awful lot, as most ATAs are quite nasty. 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
[asterisk-users] CDR duration/billsec
Hello list,
I'm having troubles implementing the ${CDR(duration)}
${CDR(billsec)} variables in this scenario:
PEER CALLS OUT -
CALL GOES TO PEER'S DEFAULT OUTGOING CONTEXT -
THE CALL IS SENT TO A MACRO AND GOES IN HANGUP -
THE CALL RETURNS TO EXTENSION h OF PEER'S DEFAULT OUTGOING CONTEXT (here I'm
trying to print the variable)
The problem is I'm always getting '0' from those variables. The
incoming calls aren't passing through a macro and they return the correct
billsec/duration value.
cdr.conf has this enabled: endbeforehexten=yes
asterisk version: 1.6.0.20
Any ideas, clues, suggestions on how may I get this to work?
Thanks in advance,
Alex
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[asterisk-users] Empty SIP Packet
Hello list, debugging SIP, I found many empty lines like: --- SIP read from UDP://XXX.XXX.XXX.XXX:5060 --- - The IP address above corresponds to one of my accounts, which is behind a firewall. Is that normal, maybe some firewall that tries to keep a port open, or is my firewall cleaning the SIP Packet? Thanks in advance best regards, Alex -- _ -- 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] Strange Problem
Hello list! I've run into a strange problem today and I was hoping that someone here has seen this before and maybe can give me a hand: I'm using asterisk 1.6.0.22 in this config: (A)PATTON ISDN -(B) ASTERISK - (C)PATTON PRI - PSTN - (D)OTHER PBX Strange Problem: USER A calls makes a call to a PBX over the PSTN and ends into an IVR. When the user makes a selection and gets his call passed to an extension of that PBX (USER D), USER D has no sound while USER A hears the voice just fine. If USER A makes a direct call to USER D, calling directly his extension, the call has audio both ways and its all working fine. The same thing if USER A calls directly mobile phones or numbers that aren't managed by IVRs. I've verified this with a few PBXs(different manufacturers), and the problem is there every time an IVR gets the control of the call. A sip debug in asterisk confirmed that the SIP Session is not renegotiated when the call exits USER's D IVR and ends up to his extension. Any idea what might be causing this? Thank you in advance! Alex -- _ -- 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] R: rtp.c:883 ast_rtcp_read: RTCP Read too short
The ring isn't fake :] The Linksys GW isn't dissing, is just responding to an INVITE. The problem is that you have problem passing voice. In other words: check RTP ports settings on server client or the firewall rules. Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di wassim darwich Inviato: giovedì 28 gennaio 2010 17:38 A: asterisk-users@lists.digium.com Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short Hi: I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys , This is wht i see on asterisk console : -- Executing [9613070...@direct:1] Set(SIP/03070741-088bd470, CALLERID(number)=96170707070) in new stack -- Executing [9613070...@direct:2] Dial(SIP/03070741-088bd470, SIP/usa/9613070741) in new stack -- Called usa/9613070741 [Jan 28 18:17:36] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short [Jan 28 18:17:42] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short -- Call on SIP/usa-08906450 left from hold -- SIP/usa-08906450 is making progress passing it to SIP/03070741-088bd470 -- SIP/usa-08906450 is ringing -- Call on SIP/usa-08906450 left from hold -- SIP/usa-08906450 is making progress passing it to SIP/03070741-088bd470 [Jan 28 18:17:50] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short [Jan 28 18:17:53] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short [Jan 28 18:17:57] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short -- _ -- 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] R: rtp.c:883 ast_rtcp_read: RTCP Read too short
Hello Wassim, server side you can check the RTP ports configured in rtp.conf which you will find in /etc/asterisk/. If the file isn't there, here are the defaults: ;[general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and rtpend=31000 You can even debug the RTP : CLI rtp debug ip xxx.xxx.xxx.xxx(linksys) Asterisk listens on one of those ports(rtp.conf ones) when a call is initiated. The same does your Linksys GW: it will listen only on the RTP configured ports. Check the firewall between the VoIP server and the Linsys GW and check the firewall on the Asterisk server. Debugging SIP you can see which ports are involved. There might be other problems, maybe because you are trying to directly pass the call from one peer(let's say an external voice provider) to the other(linksys). In that case careinvite=no is be your friend. Regards, Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di wassim darwich Inviato: giovedì 28 gennaio 2010 21:41 A: asterisk-users@lists.digium.com Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short Hi: Firewall is disabled ,so no need to worry about firewall,but i dont know where to check rtp settings and what do i need to search for ,can you guide me 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
[asterisk-users] R: How to dial multiple extensions at once likeinaring group and put them in conference?
Even though certain things should be discussed in private and that certain things should require a second thinking before stating them, I don't think you should impose a limit. Are you a Digium guy? An asterisk developer? Who are you? Apart the informational value of these lists I should hope there is a human side too. I vote for freedom of speech... Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Zeeshan Zakaria Inviato: venerdì 30 ottobre 2009 2.17 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference? Please don't make this email thread a place for political discussion. Reply only if you have any other ideas on how to accomplish what was asked in the beginning. Thanks for your understanding. -- Zeeshan A Zakaria On Thu, Oct 29, 2009 at 5:59 PM, Paul Hales pdha...@optusnet.com.aumailto:pdha...@optusnet.com.au wrote: On 29/10/09 22:40, Matt Riddell wrote: :D I should hope not!! If everyone was as smart as me, how would I take over the world? With violence, just like everyone else! PaulH ___ -- 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] R: R: R: CDR(billsec)
Thank you! My bad,the CDR function was working on 1.4, I can confirm that
endbeforehexten=yes does the trick, I've just tried it :]
WOW: On 29/10/09 5:56 AM, Alexandru Oniciuc wrote: 12h difference!
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Matt Riddell
Inviato: giovedì 29 ottobre 2009 12.44
A: asterisk-users@lists.digium.com
Oggetto: Re: [asterisk-users] R: R: CDR(billsec)
On 29/10/09 5:56 AM, Alexandru Oniciuc wrote:
I used 1.4.21 and this(${CDR(duration)}) didn't work:
exten = h,1,Verbose( (${CDR(dst)}) # Call from ${CDR(clid)} ended at
${STRFTIME(${EPOCH},,%d/%m/%Y %H:%M:%S)}. Duration(sec): ${CDR(duration)}.)
Make sure you have endbeforehexten set to yes in /etc/asterisk/cdr.conf
; Normally, CDR's are not closed out until after all extensions are
; finished executing. By enabling this option, the CDR will be ended
; before executing the h extension so that CDR values such as end
; and billsec may be retrieved inside of of this extension.
endbeforehexten=yes
--
Cheers,
Matt Riddell
Director
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[asterisk-users] R: R: R: R: CDR(billsec)
Hi Danny,
I'm using CSV output too. Maybe you didn't module reload cdr after adding
endbeforehexten?
Alex
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Danny Nicholas
Inviato: giovedì 29 ottobre 2009 14.25
A: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Oggetto: Re: [asterisk-users] R: R: R: CDR(billsec)
While I don't question that this works, I'll state that for some reason it
doesn't work on my 1.4.26.2 on Centos 5.3
Here's my test snippet
exten = 333,1,answer
exten = 333,n,SetMusicOnHold(default)
exten = 333,n,Background(pls-hold-while-try)
exten = 333,n,WaitMusicOnHold(5)
exten = 333,n,Background(vm-goodbye)
exten = 333,n,Verbose(time ${CDR(billsec)})
exten = 333,n,Verbose(dur ${CDR(duration)})
exten = 333,n,Verbose(id ${CDR(uniqueid)})
exten = 333,n,Hangup
here's cdr.conf
[general]
endbeforehexten = yes
[csv]
usegmtime = no; log date/time in GMT. Default is no
loguniqueid = yes ; log uniqueid. Default is no
loguserfield = yes ; log user field. Default is no
here's CLI output
-- Executing [...@dlpn_dialplan1:1] Answer(SIP/170-b3704ae0, ) in new stack
-- Executing [...@dlpn_dialplan1:2] SetMusicOnHold(SIP/170-b3704ae0,
default) in new stack
-- Executing [...@dlpn_dialplan1:3] BackGround(SIP/170-b3704ae0,
pls-hold-while-try) in new stack
-- SIP/170-b3704ae0 Playing 'pls-hold-while-try' (language 'en')
-- Executing [...@dlpn_dialplan1:4] WaitMusicOnHold(SIP/170-b3704ae0,
5) in new stack
-- Started music on hold, class 'default', on SIP/170-b3704ae0
-- Stopped music on hold on SIP/170-b3704ae0
-- Executing [...@dlpn_dialplan1:5] BackGround(SIP/170-b3704ae0,
vm-goodbye) in new stack
-- SIP/170-b3704ae0 Playing 'vm-goodbye' (language 'en')
-- Executing [...@dlpn_dialplan1:6] Verbose(SIP/170-b3704ae0, time 0)
in new stack
time 0
-- Executing [...@dlpn_dialplan1:7] Verbose(SIP/170-b3704ae0, dur 0) in
new stack
dur 0
-- Executing [...@dlpn_dialplan1:8] Verbose(SIP/170-b3704ae0, id
1256822108.6) in new stack
id 1256822108.6
-- Executing [...@dlpn_dialplan1:9] Hangup(SIP/170-b3704ae0, ) in new
stack
== Spawn extension (DLPN_DialPlan1, 333, 9) exited non-zero on
'SIP/170-b3704ae0'
-- Executing [...@dlpn_dialplan1:1] Set(SIP/170-b3704ae0,
CDR(userfield)= Hangupcause:16) in new stack
-- Executing [...@dlpn_dialplan1:2] DeadAGI(SIP/170-b3704ae0,
userfield.agi|1256822108.6| Hangupcause:16) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/userfield.agi
-- AGI Script userfield.agi completed, returning 0
-- Executing [...@dlpn_dialplan1:3] NoOp(SIP/170-b3704ae0, id
1256822108.6 time 8) in new stack
-- Executing [...@dlpn_dialplan1:4] NoOp(SIP/170-b3704ae0, caller hung
up) in new stack
-- Executing [...@dlpn_dialplan1:5] Hangup(SIP/170-b3704ae0, ) in new
stack
== Spawn extension (DLPN_DialPlan1, h, 5) exited non-zero on
'SIP/170-b3704ae0'
FWIW, I'm only using the csv CDR; perhaps these values are better
preserved/presented if you use the SQL CDR's?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Thursday, October 29, 2009 7:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] R: R: R: CDR(billsec)
Thanks Matt!
It works now!
Bye...
Anahi Ludueña
Date: Fri, 30 Oct 2009 01:20:02 +1300
From: li...@venturevoip.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] R: R: R: CDR(billsec)
On 30/10/09 1:10 AM, Alexandru Oniciuc wrote:
Thank you! My bad,the CDR function was working on 1.4, I can confirm that
endbeforehexten=yes does the trick, I've just tried it :]
WOW: On 29/10/09 5:56 AM, Alexandru Oniciuc wrote: 12h difference!
:D Yeah based in New Zealand - we're just about ahead of everybody - in
fact it's 1:20 in the morning so I probably should go to sleep :)
--
Cheers,
Matt Riddell
Director
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[asterisk-users] R: R: R: R: R: CDR(billsec)
Ye, don't mind that one ...
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Alexandru Oniciuc
Inviato: giovedì 29 ottobre 2009 15.15
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: [asterisk-users] R: R: R: R: CDR(billsec)
Hi Danny,
I'm using CSV output too. Maybe you didn't module reload cdr after adding
endbeforehexten?
Alex
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Danny Nicholas
Inviato: giovedì 29 ottobre 2009 14.25
A: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Oggetto: Re: [asterisk-users] R: R: R: CDR(billsec)
While I don't question that this works, I'll state that for some reason it
doesn't work on my 1.4.26.2 on Centos 5.3
Here's my test snippet
exten = 333,1,answer
exten = 333,n,SetMusicOnHold(default)
exten = 333,n,Background(pls-hold-while-try)
exten = 333,n,WaitMusicOnHold(5)
exten = 333,n,Background(vm-goodbye)
exten = 333,n,Verbose(time ${CDR(billsec)})
exten = 333,n,Verbose(dur ${CDR(duration)})
exten = 333,n,Verbose(id ${CDR(uniqueid)})
exten = 333,n,Hangup
here's cdr.conf
[general]
endbeforehexten = yes
[csv]
usegmtime = no; log date/time in GMT. Default is no
loguniqueid = yes ; log uniqueid. Default is no
loguserfield = yes ; log user field. Default is no
here's CLI output
-- Executing [...@dlpn_dialplan1:1] Answer(SIP/170-b3704ae0, ) in new stack
-- Executing [...@dlpn_dialplan1:2] SetMusicOnHold(SIP/170-b3704ae0,
default) in new stack
-- Executing [...@dlpn_dialplan1:3] BackGround(SIP/170-b3704ae0,
pls-hold-while-try) in new stack
-- SIP/170-b3704ae0 Playing 'pls-hold-while-try' (language 'en')
-- Executing [...@dlpn_dialplan1:4] WaitMusicOnHold(SIP/170-b3704ae0,
5) in new stack
-- Started music on hold, class 'default', on SIP/170-b3704ae0
-- Stopped music on hold on SIP/170-b3704ae0
-- Executing [...@dlpn_dialplan1:5] BackGround(SIP/170-b3704ae0,
vm-goodbye) in new stack
-- SIP/170-b3704ae0 Playing 'vm-goodbye' (language 'en')
-- Executing [...@dlpn_dialplan1:6] Verbose(SIP/170-b3704ae0, time 0)
in new stack
time 0
-- Executing [...@dlpn_dialplan1:7] Verbose(SIP/170-b3704ae0, dur 0) in
new stack
dur 0
-- Executing [...@dlpn_dialplan1:8] Verbose(SIP/170-b3704ae0, id
1256822108.6) in new stack
id 1256822108.6
-- Executing [...@dlpn_dialplan1:9] Hangup(SIP/170-b3704ae0, ) in new
stack
== Spawn extension (DLPN_DialPlan1, 333, 9) exited non-zero on
'SIP/170-b3704ae0'
-- Executing [...@dlpn_dialplan1:1] Set(SIP/170-b3704ae0,
CDR(userfield)= Hangupcause:16) in new stack
-- Executing [...@dlpn_dialplan1:2] DeadAGI(SIP/170-b3704ae0,
userfield.agi|1256822108.6| Hangupcause:16) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/userfield.agi
-- AGI Script userfield.agi completed, returning 0
-- Executing [...@dlpn_dialplan1:3] NoOp(SIP/170-b3704ae0, id
1256822108.6 time 8) in new stack
-- Executing [...@dlpn_dialplan1:4] NoOp(SIP/170-b3704ae0, caller hung
up) in new stack
-- Executing [...@dlpn_dialplan1:5] Hangup(SIP/170-b3704ae0, ) in new
stack
== Spawn extension (DLPN_DialPlan1, h, 5) exited non-zero on
'SIP/170-b3704ae0'
FWIW, I'm only using the csv CDR; perhaps these values are better
preserved/presented if you use the SQL CDR's?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Thursday, October 29, 2009 7:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] R: R: R: CDR(billsec)
Thanks Matt!
It works now!
Bye...
Anahi Ludueña
Date: Fri, 30 Oct 2009 01:20:02 +1300
From: li...@venturevoip.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] R: R: R: CDR(billsec)
On 30/10/09 1:10 AM, Alexandru Oniciuc wrote:
Thank you! My bad,the CDR function was working on 1.4, I can confirm that
endbeforehexten=yes does the trick, I've just tried it :]
WOW: On 29/10/09 5:56 AM, Alexandru Oniciuc wrote: 12h difference!
:D Yeah based in New Zealand - we're just about ahead of everybody - in
fact it's 1:20 in the morning so I probably should go to sleep :)
--
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
___
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[asterisk-users] R: CDR(billsec)
Hello Anahi,
I've encountered issues with CDR function when I was using the
1.4 version and was trying to get ${CDR(duration)} in extension h.
Passing to 1.6.X.X resolved it.
I hope this helps.
Alex
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Wednesday, October 28, 2009 6:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] CDR(billsec)
Hi people, when I try to get the billsec in the dialplan, it is 0... but if
after that I check the database, it is right (not 0).
I'm trying to get it in the h extension, like:
exten = h,1,Noop(End)
exten = h,n,Noop(Time is ${CDR(billsec)})
Is it updated after the extension h is executed? In that case, how can I get
the call duration in the h extension?
Thanks,
Anahi Ludueña
Todo el espacio y cuidado que merecen tus fotos digitales lo tienes en Windows
Live Fotos. ¡Pruébalo! http://www.vivelive.com/compartirfotos/
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[asterisk-users] R: R: CDR(billsec)
I used 1.4.21 and this(${CDR(duration)}) didn't work:
exten = h,1,Verbose( (${CDR(dst)}) # Call from ${CDR(clid)} ended at
${STRFTIME(${EPOCH},,%d/%m/%Y %H:%M:%S)}. Duration(sec): ${CDR(duration)}.)
Alex
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Ishfaq Malik
Inviato: mercoledì 28 ottobre 2009 17.32
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] R: CDR(billsec)
I have used ${CDR(billsec)} in asterisk 1.4.17
How I used it was
h,1,SET(BILLTIME=${CDR(billsec)})
h,2,DeadAGI(hangup.php)
My DeadAGI script could use my BILLSEC variable and it was always
consistent with the CDR too.
Danny Nicholas wrote:
Does this mean it's a bug in 1.4 or an enhancement in 1.6? If the
latter, can the change be back-ported?
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Alexandru Oniciuc
*Sent:* Wednesday, October 28, 2009 8:52 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] R: CDR(billsec)
Hello Anahi,
I've encountered issues with CDR function when I was using the 1.4
version and was trying to get ${CDR(duration)} in extension h.
Passing to 1.6.X.X resolved it.
I hope this helps.
Alex
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Anahi
Ludueña
*Sent:* Wednesday, October 28, 2009 6:35 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] CDR(billsec)
Hi people, when I try to get the billsec in the dialplan, it is 0...
but if after that I check the database, it is right (not 0).
I'm trying to get it in the h extension, like:
exten = h,1,Noop(End)
exten = h,n,Noop(Time is ${CDR(billsec)})
Is it updated after the extension h is executed? In that case, how can
I get the call duration in the h extension?
Thanks,
**
**
**Anahi Ludueña**
Todo el espacio y cuidado que merecen tus fotos digitales lo tienes en
Windows Live Fotos. ¡Pruébalo! http://www.vivelive.com/compartirfotos/
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--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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[asterisk-users] Strange IAX2 / Iaxmodem problem
Hello. I'm having a strange problem with the IAX2 channel and IAXmodem and I was hoping to get some light from someone in these lists. On my logs and on the console I'm getting a bunch of lines with: [Oct 23 14:26:18] NOTICE[4417] chan_iax2.c: Peer 'XXX' is now UNREACHABLE! Time: 3 [Oct 23 14:26:28] NOTICE[4413] chan_iax2.c: Peer 'XXX' is now REACHABLE! Time: 3 The strangeness comes when I debug the iax2 channel (iax2 set debug): the events(missed PONG I guess) that lead to the UNREACHABLE state don't appear on the console and on the log I get only 10% of the total state change messages I was getting first. Disabling the debug, the lines reappear as before. The state is changed from UNREACHABLE to REACHABLE after 10 seconds(given by qualifyfreqnotok), as documented above. Here is what I'm using to do my tests: Debian 5.0.2 Kernel 2.6.26 Asterisk 1.4.21 (tried 1.6 but the problem it's still there) Iaxmodem-1.1.1 DAHDI 2.2.0 Sangoma A104 CARD and to be safe I loaded the dahdi_dummy too *CLI dahdi show status Description Alarms IRQbpviol CRC4 wanpipe1 card 0 OK 0 0 0 wanpipe2 card 1 RED0 0 0 wanpipe3 card 2 RED0 0 0 wanpipe4 card 3 RED0 0 0 DAHDI_DUMMY/1 (source: HRtimer) 1UNCONFIGUR 0 0 0 The unreachable part is rather odd given that the two apps are on the same host. Reading some articles on the subject I found that this behavior can become a problem if the call arrives when the iaxmodem peer is in the unreachable state. Is there anyone that has an idea on what might be causing this? Sorry for the English and thanks in advance for any help you can provide, Alex ___ -- 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
