Re: [asterisk-users] how to add-edit-delete entery into asterisk conf files
Hi Virendra, That's great could you please share the sample for sip.conf and extensions.conf? On Sat, Sep 3, 2011 at 10:09 AM, virendra bhati virbh...@gmail.com wrote: Hi, I know that by using vi editor we can edit all the Linux files but I want to use Php. So that from web page anyone can make some account into asterisk server. But thanks for your reply. And i have completed that task yesterday after sending e-mail. On Sat, Sep 3, 2011 at 12:53 AM, C F shma...@gmail.com wrote: Why php? Isn't vi the only way? On Fri, Sep 2, 2011 at 7:28 AM, virendra bhati virbh...@gmail.com wrote: Hi list, I want ot do basic work (add-edit-delete) into asterisk configuration files, like sip.conf, manager.conf,musiconhold.conf etc. Please guide me how to configure all these files from from AMI connection. I am able to login into AMI from Login action but I want to do more task in to it. AMI login:- login.php ?php $socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30); fputs($socket, Action: Login\r\n); fputs($socket, UserName: root\r\n); fputs($socket, Secret: energy\r\n\r\n); ? AMI command:- Below commands are for musiconhold.conf. I want to add new MOH context into it. ?php include(login.php); fputs($socket, Action: UpdateConfig\r\n); fputs($socket, Filename: musiconhold.conf\r\n); fputs($socket, Srcfilename: musiconhold.conf\r\n); fputs($socket, Dstfilename: musiconhold.conf\r\n); fputs($socket, Action-00: newcat\r\n); fputs($socket, Cat-00: bhavik\r\n); fputs($socket, mode: files\r\n); fputs($socket, directory: /var/lib/asterisk/moh\r\n); fputs($socket, Reload: yes\r\n); fputs($socket, ActionID: 9873497149817\r\n); fputs($socket, Action: Logoff\r\n\r\n); ? After doing all no success :(( - Thanks and regards Virendra Bhati +91-9172341457 Software 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Software 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 -- Regards, Zohair Raza www.zuhair.info *http://pk.linkedin.com/in/zuhairraza** *** -- _ -- 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] how to add-edit-delete entery into asterisk conf files
ok. Thanks On Sat, Sep 3, 2011 at 5:38 PM, virendra bhati virbh...@gmail.com wrote: Hi Raza, Thanks , but there is no ned of Sip.conf and extensions.conf files. As Daniel refered the web page which is enough for all the tasks On Sat, Sep 3, 2011 at 5:18 PM, Daniel Tryba dan...@tryba.nl wrote: On Fri, Sep 02, 2011 at 04:58:52PM +0530, virendra bhati wrote: Please guide me how to configure all these files from from AMI connection. I am able to login into AMI from Login action but I want to do more task in to it. [lots of fputs] After doing all no success :(( Have you actually tried reading from the socket to see what the results are for your commands (hint: turn off events)? This is what I get: Response: Success ActionID: 9873497149817 Looking at http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+UpdateConfig it is clear that adding values to a category works different than you expected. First you need to create the new category (your scripts does that already), then you need to append lines to is in the form of appends to the category like in the example on voip-info.org: action:updateconfig reload:yes srcfilename:manager.conf dstfilename:manager.conf Action-00:append Cat-00:newuser Var-00:secret Value-00:nottelling -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Software 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 -- Regards, Zohair Raza www.zuhair.info *http://pk.linkedin.com/in/zuhairraza** *** -- _ -- 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 END TIME in correct in 1.8+
Hi, I've tested 1.8.6.0, 1.8.4.0 and 1.8.0 I can get proper start and answer time but not the end time of call SIP/11-AGI Rx GET VARIABLE CDR(start) SIP/11-AGI Tx 200 result=1 (2011-12-16 18:34:48) SIP/11-AGI Rx GET VARIABLE CDR(end) SIP/11-AGI Tx 200 result=1 (2011 12-16 18:34:48) SIP/11-AGI Rx GET VARIABLE CDR(answer) SIP/11-AGI Tx 200 result=1 (2011-12-16 18:34:50) In 1.8.6.0, there was no end time and in the other two it's present but neither in correct format nor exact time. Is it something related to system or a bug? Regards, Zohair Raza -- _ -- 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] CDR END TIME in correct in 1.8+
Yes running from h exten = _X.,1,Dial(SIP/1*100) exten = h,1,AGI(cdr.php,11) Regards, Zohair Raza On Fri, Dec 16, 2011 at 6:42 PM, Danny Nicholas da...@debsinc.com wrote: You are running the AGI from the h() exten? Otherwise I wouldn’t expect CDR(end) to populated or correct. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zohair Raza *Sent:* Friday, December 16, 2011 8:38 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] CDR END TIME in correct in 1.8+ ** ** Hi, ** ** I've tested 1.8.6.0, 1.8.4.0 and 1.8.0 ** ** I can get proper start and answer time but not the end time of call ** ** SIP/11-AGI Rx GET VARIABLE CDR(start) SIP/11-AGI Tx 200 result=1 (2011-12-16 18:34:48) SIP/11-AGI Rx GET VARIABLE CDR(end) SIP/11-AGI Tx 200 result=1 (2011 12-16 18:34:48) SIP/11-AGI Rx GET VARIABLE CDR(answer) SIP/11-AGI Tx 200 result=1 (2011-12-16 18:34:50) ** ** In 1.8.6.0, there was no end time and in the other two it's present but neither in correct format nor exact time. ** ** Is it something related to system or a bug? ** ** Regards, Zohair Raza ** ** -- _ -- 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] CDR END TIME in correct in 1.8+
Still same, even when I am trying to write in one agi and calling it using DeadAGI Regards, Zohair Raza On Fri, Dec 16, 2011 at 6:56 PM, Danny Nicholas da...@debsinc.com wrote: Try this exten = _X.,1,Dial(SIP/1*100) exten = h,1,wait(10) exten = h,n,AGI(cdr.php,11) ** ** Don’t know how long after hangup this information gets updated, but would be shocked if 10 seconds doesn’t cover it. ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zohair Raza *Sent:* Friday, December 16, 2011 8:51 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] CDR END TIME in correct in 1.8+ ** ** Yes running from h ** ** ** ** exten = _X.,1,Dial(SIP/1*100) exten = h,1,AGI(cdr.php,11) ** ** Regards, Zohair Raza ** ** On Fri, Dec 16, 2011 at 6:42 PM, Danny Nicholas da...@debsinc.com wrote: You are running the AGI from the h() exten? Otherwise I wouldn’t expect CDR(end) to populated or correct. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zohair Raza *Sent:* Friday, December 16, 2011 8:38 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] CDR END TIME in correct in 1.8+ Hi, I've tested 1.8.6.0, 1.8.4.0 and 1.8.0 I can get proper start and answer time but not the end time of call SIP/11-AGI Rx GET VARIABLE CDR(start) SIP/11-AGI Tx 200 result=1 (2011-12-16 18:34:48) SIP/11-AGI Rx GET VARIABLE CDR(end) SIP/11-AGI Tx 200 result=1 (2011 12-16 18:34:48) SIP/11-AGI Rx GET VARIABLE CDR(answer) SIP/11-AGI Tx 200 result=1 (2011-12-16 18:34:50) In 1.8.6.0, there was no end time and in the other two it's present but neither in correct format nor exact time. Is it something related to system or a bug? Regards, Zohair Raza -- _ -- 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 -- _ -- 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] CDR END TIME in correct in 1.8+
thanks, It worked for h! and if I want in DeadAGI? I want cdr function in the same AGI. Regards, Zohair Raza On Fri, Dec 16, 2011 at 7:08 PM, Eric Wieling ewiel...@nyigc.com wrote: From cdr.conf.sample: ; 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=no -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, December 16, 2011 9:57 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] CDR END TIME in correct in 1.8+ Try this exten = _X.,1,Dial(SIP/1*100) exten = h,1,wait(10) exten = h,n,AGI(cdr.php,11) Don't know how long after hangup this information gets updated, but would be shocked if 10 seconds doesn't cover it. From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Zohair Raza Sent: Friday, December 16, 2011 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR END TIME in correct in 1.8+ Yes running from h exten = _X.,1,Dial(SIP/1*100) exten = h,1,AGI(cdr.php,11) Regards, Zohair Raza On Fri, Dec 16, 2011 at 6:42 PM, Danny Nicholas da...@debsinc.com wrote: You are running the AGI from the h() exten? Otherwise I wouldn't expect CDR(end) to populated or correct. From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Zohair Raza Sent: Friday, December 16, 2011 8:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CDR END TIME in correct in 1.8+ Hi, I've tested 1.8.6.0, 1.8.4.0 and 1.8.0 I can get proper start and answer time but not the end time of call SIP/11-AGI Rx GET VARIABLE CDR(start) SIP/11-AGI Tx 200 result=1 (2011-12-16 18:34:48) SIP/11-AGI Rx GET VARIABLE CDR(end) SIP/11-AGI Tx 200 result=1 (2011 12-16 18:34:48) SIP/11-AGI Rx GET VARIABLE CDR(answer) SIP/11-AGI Tx 200 result=1 (2011-12-16 18:34:50) In 1.8.6.0, there was no end time and in the other two it's present but neither in correct format nor exact time. Is it something related to system or a bug? Regards, Zohair Raza -- _ -- 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 -- _ -- 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] How to monitor SIP Trunk on production server
Hi, http://blog.tmcnet.com/blog/tom-keating/asterisk/using-monit-tool-to-monitor-asterisk.asp Regards, Zohair Raza On Sun, Dec 18, 2011 at 9:26 AM, virendra bhati virbh...@gmail.com wrote: Hi List, I have asterisk 1.6.2.20 installed at production server, I have 2 SIP voip trunk for making outgoing and DID for incoming to server. My question is how I can ensure that trunk is not down at production server, So how I can monitor it's automatically by making any scripts? Any hint will be appreciated -- Thanks and regards Virendra Bhati +91-8885268942 Software 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 -- _ -- 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] Called peer IP
Hi List,
Which will be the appropriate variable to get called peer IP address?
I tried following channel variables
peerip, recvip, URI, from
and following SIP channel variables:
SIPURI,SIPDOMAIN
They all return calling peer IP but not the destination/called peer IP.
unfortunately set(CDR(calledip)=${CHANNEL(to)}) doesn't work
Regards,
Zohair Raza
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Re: [asterisk-users] cdr call time
may this helps, In cdr.conf, set endbeforehexten=yes Regards, Zohair Raza On Wed, Dec 28, 2011 at 4:46 AM, Vinod Dharashive vdharash...@gmail.comwrote: Hi team, On event of no answer in CDR the starttime and endtime of call remains the same. Is there any way how can actually track call originate time and call end time. Thanks Vinod dharashive. Sent from BlackBerry® on Airtel -- _ -- 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] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
all of them have a wiki page http://lmgtfy.com/?q=Asterisk http://lmgtfy.com/?q=freeswitch http://lmgtfy.com/?q=openser http://lmgtfy.com/?q=TrixBox Regards, Zohair Raza On Tue, Jan 3, 2012 at 5:47 PM, Kaushal Shriyan kaushalshri...@gmail.comwrote: Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -- _ -- 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] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Hi, This may help you. http://www.techistan.com/2010/05/31/difference-between-kamailio-and-freeswitch-or-asterisk-and-more-with-mierla/ Regards, Zohair Raza On Tue, Jan 3, 2012 at 5:57 PM, Kaushal Shriyan kaushalshri...@gmail.comwrote: On Tue, Jan 3, 2012 at 7:23 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: all of them have a wiki page http://lmgtfy.com/?q=Asterisk http://lmgtfy.com/?q=freeswitch http://lmgtfy.com/?q=openser http://lmgtfy.com/?q=TrixBox Regards, Zohair Raza Hi Zohair I was interested in some sort of comparison sheet and its advantages over each other. Regards Kaushal -- _ -- 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] DIALSTATUS Values
This works fine for me, $dialstatus = $agi-get_variable(DIALSTATUS); $cdr['dialstatus'] = $dialstatus['data']; Try as it is, I believe it's because of concatenation. Regards, Zohair Raza On Fri, Dec 2, 2011 at 4:27 PM, Tony Mountifield t...@softins.co.uk wrote: In article snt142-w54267269808afd17bccd5891...@phx.gbl, Kamlesh Kumar kamlesh_...@hotmail.com wrote: In addition to my reply: I used to fetch the value using print_r function but that also tells that there is no value in data section. $dialstatus=$agi-get_variable(DIALSTATUS); print_r($dialstatus); SIP/10036-00b8AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b8AGI Tx 200 result=1 (CANCEL) SIP/10036-00b8AGI Rx Array SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx ( SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [code] = 200 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [result] = 1 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [data] = Well since the AGI return string does indeed contain the value, shown above as (CANCEL), that suggests there is definitely a bug in php-agi. It appears to be creating a ['data'] element, but not setting it. You will have to study the source code and work out how to fix it. I did a quick google for php agi get variable and found other reports of it not working properly, but I didn't see anyone offer a solution. It's only programming, so it shouldn't be hard to fix. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set callerid in php AGI file.
Hi,
Try setting CDR(clid)
Regards,
Zohair Raza
On Thu, Jan 12, 2012 at 12:44 PM, virendra bhati virbh...@gmail.com wrote:
Hi,
I am using phpagi for agi scripting. and want to update callerid number
but didn't get any success. please help me how to update PHPAGI is new for
me. Below is the code which I write.
#!/usr/bin/php -q
?php
set_time_limit(30);
//require(.phpagi.php.);
include(phpagi.php);
$agi = new AGI();
//answer the call
$agi- answer();
$agi-verbose(--);
$agi- exec('Set',CALLERID(num)=01133200274);
$ani = $agi-request['agi_callerid'];
$agi-noop(My CalleID: =.$ani);
$agi-set_variable(CALLERID(num),01133200274);
$ani = $agi-request['agi_callerid'];
$agi-noop(My CalleID: =.$ani);
$agi- exec('Dial',SIP/00918885268...@sip.trunk.gradwell.com,60,r);
//$agi- exec('Dial',SIP/00918885268942@voipon,60,r);
?
And CLI
== Using SIP RTP CoS mark 5
-- Executing [101@outbound:1] Answer(SIP/2209-26d3, ) in new
stack
-- Executing [101@outbound:2] AGI(SIP/2209-26d3,
/home/virendra.bhati/outdial.php) in new stack
-- Launched AGI Script /home/virendra.bhati/outdial.php
SIP/2209-26d3AGI Tx agi_request: /home/virendra.bhati/outdial.php
SIP/2209-26d3AGI Tx agi_channel: SIP/2209-26d3
SIP/2209-26d3AGI Tx agi_language: en
SIP/2209-26d3AGI Tx agi_type: SIP
SIP/2209-26d3AGI Tx agi_uniqueid: 1326357644.10070
SIP/2209-26d3AGI Tx agi_version: 1.6.2.20
SIP/2209-26d3AGI Tx agi_callerid: 2209
SIP/2209-26d3AGI Tx agi_calleridname: unknown
SIP/2209-26d3AGI Tx agi_callingpres: 0
SIP/2209-26d3AGI Tx agi_callingani2: 0
SIP/2209-26d3AGI Tx agi_callington: 0
SIP/2209-26d3AGI Tx agi_callingtns: 0
SIP/2209-26d3AGI Tx agi_dnid: 101
SIP/2209-26d3AGI Tx agi_rdnis: unknown
SIP/2209-26d3AGI Tx agi_context: outbound
SIP/2209-26d3AGI Tx agi_extension: 101
SIP/2209-26d3AGI Tx agi_priority: 2
SIP/2209-26d3AGI Tx agi_enhanced: 0.0
SIP/2209-26d3AGI Tx agi_accountcode:
SIP/2209-26d3AGI Tx agi_threadid: 1386719552
SIP/2209-26d3AGI Tx
SIP/2209-26d3AGI Rx ANSWER
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx VERBOSE
-- 1
/home/virendra.bhati/outdial.php:
--
SIP/2209-26d3AGI Tx 200 result=1
SIP/2209-26d3AGI Rx EXEC Set CALLERID(num)=01133200274
-- AGI Script Executing Application: (Set) Options:
(CALLERID(num)=01133200274)
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx NOOP My CalleID: =2209
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx SET VARIABLE CALLERID(num) 01133200274
SIP/2209-26d3AGI Tx 200 result=1
SIP/2209-26d3AGI Rx NOOP My CalleID: =2209
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx EXEC Dial SIP/
00918885268...@sip.trunk.gradwell.com,60,r
-- AGI Script Executing Application: (Dial) Options: (SIP/
00918885268...@sip.trunk.gradwell.com,60,r)
== Using SIP RTP CoS mark 5
ast_get_srv: SRV lookup for '_sip._udp.sip.trunk.gradwell.com'
mapped to host v-sip-trunk-out-f1.gradwell.net, port 5060
-- Called 00918885268...@sip.trunk.gradwell.com
[Jan 12 14:10:52] WARNING[28001]: chan_sip.c:18463 handle_response_invite:
Received response: Forbidden from '01133200274
sip:01133200274@10.10.10.181;tag=as76229e88'
-- SIP/sip.trunk.gradwell.com-26d4 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
SIP/2209-26d3AGI Tx 200 result=0
-- SIP/2209-26d3AGI Script /home/virendra.bhati/outdial.php
completed, returning 0
-- Executing [101@outbound:3] Hangup(SIP/2209-26d3, ) in new
stack
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
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Re: [asterisk-users] how to set callerid in php AGI file.
In phpagi
$agi-set_variable(CDR(clid) )
and to get it
$agi-get_variable(CDR(clid))
Regards,
Zohair Raza
www.zuhair.info
*http://ae.linkedin.com/in/zuhairraza** ***
On Thu, Jan 12, 2012 at 12:51 PM, virendra bhati virbh...@gmail.com wrote:
How to used it in AGI ? I think it's Dialplan apps.
On Thu, Jan 12, 2012 at 2:18 PM, Zohair Raza engineerzuhairr...@gmail.com
wrote:
Hi,
Try setting CDR(clid)
Regards,
Zohair Raza
On Thu, Jan 12, 2012 at 12:44 PM, virendra bhati virbh...@gmail.comwrote:
Hi,
I am using phpagi for agi scripting. and want to update callerid number
but didn't get any success. please help me how to update PHPAGI is new for
me. Below is the code which I write.
#!/usr/bin/php -q
?php
set_time_limit(30);
//require(.phpagi.php.);
include(phpagi.php);
$agi = new AGI();
//answer the call
$agi- answer();
$agi-verbose(--);
$agi- exec('Set',CALLERID(num)=01133200274);
$ani = $agi-request['agi_callerid'];
$agi-noop(My CalleID: =.$ani);
$agi-set_variable(CALLERID(num),01133200274);
$ani = $agi-request['agi_callerid'];
$agi-noop(My CalleID: =.$ani);
$agi- exec('Dial',SIP/00918885268...@sip.trunk.gradwell.com,60,r);
//$agi- exec('Dial',SIP/00918885268942@voipon,60,r);
?
And CLI
== Using SIP RTP CoS mark 5
-- Executing [101@outbound:1] Answer(SIP/2209-26d3, ) in
new stack
-- Executing [101@outbound:2] AGI(SIP/2209-26d3,
/home/virendra.bhati/outdial.php) in new stack
-- Launched AGI Script /home/virendra.bhati/outdial.php
SIP/2209-26d3AGI Tx agi_request:
/home/virendra.bhati/outdial.php
SIP/2209-26d3AGI Tx agi_channel: SIP/2209-26d3
SIP/2209-26d3AGI Tx agi_language: en
SIP/2209-26d3AGI Tx agi_type: SIP
SIP/2209-26d3AGI Tx agi_uniqueid: 1326357644.10070
SIP/2209-26d3AGI Tx agi_version: 1.6.2.20
SIP/2209-26d3AGI Tx agi_callerid: 2209
SIP/2209-26d3AGI Tx agi_calleridname: unknown
SIP/2209-26d3AGI Tx agi_callingpres: 0
SIP/2209-26d3AGI Tx agi_callingani2: 0
SIP/2209-26d3AGI Tx agi_callington: 0
SIP/2209-26d3AGI Tx agi_callingtns: 0
SIP/2209-26d3AGI Tx agi_dnid: 101
SIP/2209-26d3AGI Tx agi_rdnis: unknown
SIP/2209-26d3AGI Tx agi_context: outbound
SIP/2209-26d3AGI Tx agi_extension: 101
SIP/2209-26d3AGI Tx agi_priority: 2
SIP/2209-26d3AGI Tx agi_enhanced: 0.0
SIP/2209-26d3AGI Tx agi_accountcode:
SIP/2209-26d3AGI Tx agi_threadid: 1386719552
SIP/2209-26d3AGI Tx
SIP/2209-26d3AGI Rx ANSWER
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx VERBOSE
-- 1
/home/virendra.bhati/outdial.php:
--
SIP/2209-26d3AGI Tx 200 result=1
SIP/2209-26d3AGI Rx EXEC Set CALLERID(num)=01133200274
-- AGI Script Executing Application: (Set) Options: (CALLERID(num)=
01133200274)
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx NOOP My CalleID: =2209
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx SET VARIABLE CALLERID(num) 01133200274
SIP/2209-26d3AGI Tx 200 result=1
SIP/2209-26d3AGI Rx NOOP My CalleID: =2209
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx EXEC Dial SIP/
00918885268...@sip.trunk.gradwell.com,60,r
-- AGI Script Executing Application: (Dial) Options: (SIP/
00918885268...@sip.trunk.gradwell.com,60,r)
== Using SIP RTP CoS mark 5
ast_get_srv: SRV lookup for '_sip._udp.sip.trunk.gradwell.com'
mapped to host v-sip-trunk-out-f1.gradwell.net, port 5060
-- Called 00918885268...@sip.trunk.gradwell.com
[Jan 12 14:10:52] WARNING[28001]: chan_sip.c:18463
handle_response_invite: Received response: Forbidden from '
01133200274 sip:01133200274@10.10.10.181;tag=as76229e88'
-- SIP/sip.trunk.gradwell.com-26d4 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
SIP/2209-26d3AGI Tx 200 result=0
-- SIP/2209-26d3AGI Script /home/virendra.bhati/outdial.php
completed, returning 0
-- Executing [101@outbound:3] Hangup(SIP/2209-26d3, ) in
new stack
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
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asterisk-users mailing list
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To UNSUBSCRIBE or update
Re: [asterisk-users] how to set callerid in php AGI file.
Any variable can be set and get from agi
CDR(clid) is a CDR variable
Regards,
Zohair Raza
On Thu, Jan 12, 2012 at 12:51 PM, virendra bhati virbh...@gmail.com wrote:
How to used it in AGI ? I think it's Dialplan apps.
On Thu, Jan 12, 2012 at 2:18 PM, Zohair Raza engineerzuhairr...@gmail.com
wrote:
Hi,
Try setting CDR(clid)
Regards,
Zohair Raza
On Thu, Jan 12, 2012 at 12:44 PM, virendra bhati virbh...@gmail.comwrote:
Hi,
I am using phpagi for agi scripting. and want to update callerid number
but didn't get any success. please help me how to update PHPAGI is new for
me. Below is the code which I write.
#!/usr/bin/php -q
?php
set_time_limit(30);
//require(.phpagi.php.);
include(phpagi.php);
$agi = new AGI();
//answer the call
$agi- answer();
$agi-verbose(--);
$agi- exec('Set',CALLERID(num)=01133200274);
$ani = $agi-request['agi_callerid'];
$agi-noop(My CalleID: =.$ani);
$agi-set_variable(CALLERID(num),01133200274);
$ani = $agi-request['agi_callerid'];
$agi-noop(My CalleID: =.$ani);
$agi- exec('Dial',SIP/00918885268...@sip.trunk.gradwell.com,60,r);
//$agi- exec('Dial',SIP/00918885268942@voipon,60,r);
?
And CLI
== Using SIP RTP CoS mark 5
-- Executing [101@outbound:1] Answer(SIP/2209-26d3, ) in
new stack
-- Executing [101@outbound:2] AGI(SIP/2209-26d3,
/home/virendra.bhati/outdial.php) in new stack
-- Launched AGI Script /home/virendra.bhati/outdial.php
SIP/2209-26d3AGI Tx agi_request:
/home/virendra.bhati/outdial.php
SIP/2209-26d3AGI Tx agi_channel: SIP/2209-26d3
SIP/2209-26d3AGI Tx agi_language: en
SIP/2209-26d3AGI Tx agi_type: SIP
SIP/2209-26d3AGI Tx agi_uniqueid: 1326357644.10070
SIP/2209-26d3AGI Tx agi_version: 1.6.2.20
SIP/2209-26d3AGI Tx agi_callerid: 2209
SIP/2209-26d3AGI Tx agi_calleridname: unknown
SIP/2209-26d3AGI Tx agi_callingpres: 0
SIP/2209-26d3AGI Tx agi_callingani2: 0
SIP/2209-26d3AGI Tx agi_callington: 0
SIP/2209-26d3AGI Tx agi_callingtns: 0
SIP/2209-26d3AGI Tx agi_dnid: 101
SIP/2209-26d3AGI Tx agi_rdnis: unknown
SIP/2209-26d3AGI Tx agi_context: outbound
SIP/2209-26d3AGI Tx agi_extension: 101
SIP/2209-26d3AGI Tx agi_priority: 2
SIP/2209-26d3AGI Tx agi_enhanced: 0.0
SIP/2209-26d3AGI Tx agi_accountcode:
SIP/2209-26d3AGI Tx agi_threadid: 1386719552
SIP/2209-26d3AGI Tx
SIP/2209-26d3AGI Rx ANSWER
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx VERBOSE
-- 1
/home/virendra.bhati/outdial.php:
--
SIP/2209-26d3AGI Tx 200 result=1
SIP/2209-26d3AGI Rx EXEC Set CALLERID(num)=01133200274
-- AGI Script Executing Application: (Set) Options: (CALLERID(num)=
01133200274)
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx NOOP My CalleID: =2209
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx SET VARIABLE CALLERID(num) 01133200274
SIP/2209-26d3AGI Tx 200 result=1
SIP/2209-26d3AGI Rx NOOP My CalleID: =2209
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx EXEC Dial SIP/
00918885268...@sip.trunk.gradwell.com,60,r
-- AGI Script Executing Application: (Dial) Options: (SIP/
00918885268...@sip.trunk.gradwell.com,60,r)
== Using SIP RTP CoS mark 5
ast_get_srv: SRV lookup for '_sip._udp.sip.trunk.gradwell.com'
mapped to host v-sip-trunk-out-f1.gradwell.net, port 5060
-- Called 00918885268...@sip.trunk.gradwell.com
[Jan 12 14:10:52] WARNING[28001]: chan_sip.c:18463
handle_response_invite: Received response: Forbidden from '
01133200274 sip:01133200274@10.10.10.181;tag=as76229e88'
-- SIP/sip.trunk.gradwell.com-26d4 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
SIP/2209-26d3AGI Tx 200 result=0
-- SIP/2209-26d3AGI Script /home/virendra.bhati/outdial.php
completed, returning 0
-- Executing [101@outbound:3] Hangup(SIP/2209-26d3, ) in
new stack
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software 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
--
_
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asterisk-users mailing list
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--
Thanks
Re: [asterisk-users] how to set callerid in php AGI file.
Phpagi also has predefined method
$agi - set_callerid();
Regards,
Zohair Raza
On Thu, Jan 12, 2012 at 1:02 PM, Zohair Raza
engineerzuhairr...@gmail.comwrote:
Any variable can be set and get from agi
CDR(clid) is a CDR variable
Regards,
Zohair Raza
On Thu, Jan 12, 2012 at 12:51 PM, virendra bhati virbh...@gmail.comwrote:
How to used it in AGI ? I think it's Dialplan apps.
On Thu, Jan 12, 2012 at 2:18 PM, Zohair Raza
engineerzuhairr...@gmail.com wrote:
Hi,
Try setting CDR(clid)
Regards,
Zohair Raza
On Thu, Jan 12, 2012 at 12:44 PM, virendra bhati virbh...@gmail.comwrote:
Hi,
I am using phpagi for agi scripting. and want to update callerid number
but didn't get any success. please help me how to update PHPAGI is new for
me. Below is the code which I write.
#!/usr/bin/php -q
?php
set_time_limit(30);
//require(.phpagi.php.);
include(phpagi.php);
$agi = new AGI();
//answer the call
$agi- answer();
$agi-verbose(--);
$agi- exec('Set',CALLERID(num)=01133200274);
$ani = $agi-request['agi_callerid'];
$agi-noop(My CalleID: =.$ani);
$agi-set_variable(CALLERID(num),01133200274);
$ani = $agi-request['agi_callerid'];
$agi-noop(My CalleID: =.$ani);
$agi- exec('Dial',SIP/00918885268...@sip.trunk.gradwell.com,60,r);
//$agi- exec('Dial',SIP/00918885268942@voipon,60,r);
?
And CLI
== Using SIP RTP CoS mark 5
-- Executing [101@outbound:1] Answer(SIP/2209-26d3, ) in
new stack
-- Executing [101@outbound:2] AGI(SIP/2209-26d3,
/home/virendra.bhati/outdial.php) in new stack
-- Launched AGI Script /home/virendra.bhati/outdial.php
SIP/2209-26d3AGI Tx agi_request:
/home/virendra.bhati/outdial.php
SIP/2209-26d3AGI Tx agi_channel: SIP/2209-26d3
SIP/2209-26d3AGI Tx agi_language: en
SIP/2209-26d3AGI Tx agi_type: SIP
SIP/2209-26d3AGI Tx agi_uniqueid: 1326357644.10070
SIP/2209-26d3AGI Tx agi_version: 1.6.2.20
SIP/2209-26d3AGI Tx agi_callerid: 2209
SIP/2209-26d3AGI Tx agi_calleridname: unknown
SIP/2209-26d3AGI Tx agi_callingpres: 0
SIP/2209-26d3AGI Tx agi_callingani2: 0
SIP/2209-26d3AGI Tx agi_callington: 0
SIP/2209-26d3AGI Tx agi_callingtns: 0
SIP/2209-26d3AGI Tx agi_dnid: 101
SIP/2209-26d3AGI Tx agi_rdnis: unknown
SIP/2209-26d3AGI Tx agi_context: outbound
SIP/2209-26d3AGI Tx agi_extension: 101
SIP/2209-26d3AGI Tx agi_priority: 2
SIP/2209-26d3AGI Tx agi_enhanced: 0.0
SIP/2209-26d3AGI Tx agi_accountcode:
SIP/2209-26d3AGI Tx agi_threadid: 1386719552
SIP/2209-26d3AGI Tx
SIP/2209-26d3AGI Rx ANSWER
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx VERBOSE
-- 1
/home/virendra.bhati/outdial.php:
--
SIP/2209-26d3AGI Tx 200 result=1
SIP/2209-26d3AGI Rx EXEC Set CALLERID(num)=01133200274
-- AGI Script Executing Application: (Set) Options: (CALLERID(num)=
01133200274)
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx NOOP My CalleID: =2209
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx SET VARIABLE CALLERID(num) 01133200274
SIP/2209-26d3AGI Tx 200 result=1
SIP/2209-26d3AGI Rx NOOP My CalleID: =2209
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx EXEC Dial SIP/
00918885268...@sip.trunk.gradwell.com,60,r
-- AGI Script Executing Application: (Dial) Options: (SIP/
00918885268...@sip.trunk.gradwell.com,60,r)
== Using SIP RTP CoS mark 5
ast_get_srv: SRV lookup for '_sip._udp.sip.trunk.gradwell.com'
mapped to host v-sip-trunk-out-f1.gradwell.net, port 5060
-- Called 00918885268...@sip.trunk.gradwell.com
[Jan 12 14:10:52] WARNING[28001]: chan_sip.c:18463
handle_response_invite: Received response: Forbidden from '
01133200274 sip:01133200274@10.10.10.181;tag=as76229e88'
-- SIP/sip.trunk.gradwell.com-26d4 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
SIP/2209-26d3AGI Tx 200 result=0
-- SIP/2209-26d3AGI Script /home/virendra.bhati/outdial.php
completed, returning 0
-- Executing [101@outbound:3] Hangup(SIP/2209-26d3, ) in
new stack
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software 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:
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_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http
[asterisk-users] Prepaid billing
Hi All, I am writing a billing engine in AGI. My scenario is : One customer can have simultaneous calls and I need to hang up one customer's all call when balance reaches 0 If I set limit for each call using 'L' in dial command, lets say 5 minutes in accordance with remaining credit and connect the call, few seconds later a 2nd call comes in and the first call is still in progress. If I permit the same 5 minutes as per this formula and both calls remains connected for the next 5 minutes then credit will go in minus which is not acceptable. One option is to charge credit via AMI and as soon as the credit goes 0, hangup all calls for this customer. Is there any other way to achieve this ? Regards, Zohair Raza -- _ -- 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] Prepaid billing
Hi, I understand this, but I think there isn't any option that helps us to reduce cost while call is in progress. One option that I was thinking is to check elapsed time by core show channel channel-id and deduct the amount but we need to check it every second or x seconds via AMI. Regards, Zohair Raza On Wed, Jan 18, 2012 at 9:35 AM, virendra bhati virbh...@gmail.com wrote: Hi Zohair, By using only asterisk it's not possible. So used progremming languages and do realtime billing at your ends. like 1st caller will take complete amount ($5) and if 2nd call will come then deduct used amount and share remaining amount to others like that. On Tue, Jan 17, 2012 at 9:54 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hi All, I am writing a billing engine in AGI. My scenario is : One customer can have simultaneous calls and I need to hang up one customer's all call when balance reaches 0 If I set limit for each call using 'L' in dial command, lets say 5 minutes in accordance with remaining credit and connect the call, few seconds later a 2nd call comes in and the first call is still in progress. If I permit the same 5 minutes as per this formula and both calls remains connected for the next 5 minutes then credit will go in minus which is not acceptable. One option is to charge credit via AMI and as soon as the credit goes 0, hangup all calls for this customer. Is there any other way to achieve this ? Regards, Zohair Raza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software 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 -- _ -- 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] Prepaid billing
Oh yes that will be more suitable but will still need to do it via AMI Regards, Zohair Raza On Wed, Jan 18, 2012 at 11:35 AM, virendra bhati virbh...@gmail.com wrote: Batter is used DB to store intime of call then when ever currect used time is required then deduct from intime - current time. On Wed, Jan 18, 2012 at 1:01 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hi, I understand this, but I think there isn't any option that helps us to reduce cost while call is in progress. One option that I was thinking is to check elapsed time by core show channel channel-id and deduct the amount but we need to check it every second or x seconds via AMI. Regards, Zohair Raza On Wed, Jan 18, 2012 at 9:35 AM, virendra bhati virbh...@gmail.comwrote: Hi Zohair, By using only asterisk it's not possible. So used progremming languages and do realtime billing at your ends. like 1st caller will take complete amount ($5) and if 2nd call will come then deduct used amount and share remaining amount to others like that. On Tue, Jan 17, 2012 at 9:54 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hi All, I am writing a billing engine in AGI. My scenario is : One customer can have simultaneous calls and I need to hang up one customer's all call when balance reaches 0 If I set limit for each call using 'L' in dial command, lets say 5 minutes in accordance with remaining credit and connect the call, few seconds later a 2nd call comes in and the first call is still in progress. If I permit the same 5 minutes as per this formula and both calls remains connected for the next 5 minutes then credit will go in minus which is not acceptable. One option is to charge credit via AMI and as soon as the credit goes 0, hangup all calls for this customer. Is there any other way to achieve this ? Regards, Zohair Raza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software 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 -- _ -- 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] Playback with noanswer in AGI
Thanks for this explanation Dany! Regards, Zohair Raza On Mon, Feb 6, 2012 at 10:11 PM, Danny Nicholas da...@debsinc.com wrote: You are mis-understanding the concept – the noanswer option is playing the file as you requested, but since you aren’t answering the call, no channel is established to actually present the sound to you. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zohair Raza *Sent:* Monday, February 06, 2012 12:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Playback with noanswer in AGI ** ** Hi All, ** ** I want to play a file in agi but dont want to answer the call ** ** I am dialing through sip phone and running asterisk 1.8.6, ** ** I tried following with no luck ** ** $agi-exec(Progress); $agi-exec(Playback $filetoplay,noanswer); $agi-hangup(); ** ** When I dial I can't hear the audio but if I answer the call or remove noanswer argument I can hear the audio. ** ** phpAGI's stream_file didn't help either. ** ** I ended up with ResetCDR() before hangup to reset billsec, duration and disposition but don't want to do it this way. ** ** What could be the problem? ** ** From Voip-info.org : *noanswer*: Play the sound file, but don't answer the channel first (if hasn't been answered already). Not all channels support playing messages while still on hook. ** ** Is it because the channel is not supported? ** ** ** ** Regards, Zohair Raza ** ** ** ** -- _ -- 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] Playback with noanswer in AGI
Hi Sammy, Thanks for input. I have an eyebeam softphone registered with Asterisk 1.8.6 locally and from agi, I pass this $filetoplay = 'congestion'; $agi-exec(Progress); $agi-exec(Playback $filetoplay,noanswer); Have tried putting file in .gsm and .wav formats, I hear ringing tone instead of playback Please have a look at sip-trace --- SIP read from UDP:176.249.0.50:8721 --- INVITE sip:100@176.249.0.77 SIP/2.0 To: sip:100@176.249.0.77 From: Zohairsip:1000@176.249.0.77;tag=7f222672 Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport Call-ID: 2932f90ef302332b CSeq: 2 INVITE Contact: sip:1000@176.249.0.50:8721 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3006o stamp 17551 Authorization: Digest username=1000,realm=asterisk,nonce=2abce759,uri=sip:100@176.249.0.77 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5 Content-Length: 269 v=0 o=- 4333518 4333604 IN IP4 176.249.0.50 s=eyeBeam c=IN IP4 176.249.0.50 t=0 0 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 11 lines) --- Sending to 176.249.0.50:8721 (no NAT) sing INVITE request as basis request - 2932f90ef302332b Found peer '1000' for '1000' from 176.249.0.50:8721 == Using SIP RTP CoS mark 5 Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 101 Found audio description format speex for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 176.249.0.50:6506 Looking for 100 in default (domain 176.249.0.77) list_route: hop: sip:1000@176.249.0.50:8721 --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Length: 0 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID) -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php -- AGI Script Executing Application: (Progress) Options: () Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77;tag=as01491743 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Type: application/sdp Content-Length: 258 v=0 o=root 1225456982 1225456982 IN IP4 176.249.0.77 s=Asterisk PBX 1.8.0 c=IN IP4 176.249.0.77 t=0 0 m=audio 15918 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- AGI Script Executing Application: (Playback) Options: (congestion,noanswer) -- SIP/1000-0019 Playing 'congestion.slin' (language 'en') -- SIP/1000-0019AGI Script agi.php completed, returning 0 Regards, Zohair Raza On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote: Hey Danny, I've this thing exactly running and working as Zohair mentioned! i.e I do not answer() the call rather put a progress() and soon after that playing back the sound file from playback with noanswer and then I get the file streaming as 183-Session progress file. I do understand that playing any sound file before establishing any audio session between two end point will result in no-adio from playback() BUT the combination of progress() and playback(,noanswer) works fine for me. What I think the issue could be for Zohair is that its requesting/incoming session(carrier) isn't allowing the 183-Session progress. Zohair can you do a SIP trace for this particular call along with the dialplan executing for it!? Regards, Sammy. On Tue, Feb 7, 2012 at 11:55 AM, Zohair Raza engineerzuhairr...@gmail.com wrote
Re: [asterisk-users] Playback with noanswer in AGI
Sammy, Problem is at phones, with a linksys phone it works but with eyebeam and fanvill it doesn't Maybe they don't support early media. I think i will have to stick with ResetCDR and that will be okay now as I've modified the code for that Thank you Regards, Zohair Raza On Tue, Feb 7, 2012 at 12:09 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: Hi Sammy, Thanks for input. I have an eyebeam softphone registered with Asterisk 1.8.6 locally and from agi, I pass this $filetoplay = 'congestion'; $agi-exec(Progress); $agi-exec(Playback $filetoplay,noanswer); Have tried putting file in .gsm and .wav formats, I hear ringing tone instead of playback Please have a look at sip-trace --- SIP read from UDP:176.249.0.50:8721 --- INVITE sip:100@176.249.0.77 SIP/2.0 To: sip:100@176.249.0.77 From: Zohairsip:1000@176.249.0.77;tag=7f222672 Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport Call-ID: 2932f90ef302332b CSeq: 2 INVITE Contact: sip:1000@176.249.0.50:8721 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3006o stamp 17551 Authorization: Digest username=1000,realm=asterisk,nonce=2abce759,uri= sip:100@176.249.0.77 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5 Content-Length: 269 v=0 o=- 4333518 4333604 IN IP4 176.249.0.50 s=eyeBeam c=IN IP4 176.249.0.50 t=0 0 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 11 lines) --- Sending to 176.249.0.50:8721 (no NAT) sing INVITE request as basis request - 2932f90ef302332b Found peer '1000' for '1000' from 176.249.0.50:8721 == Using SIP RTP CoS mark 5 Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 101 Found audio description format speex for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 176.249.0.50:6506 Looking for 100 in default (domain 176.249.0.77) list_route: hop: sip:1000@176.249.0.50:8721 --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Length: 0 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID) -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php -- AGI Script Executing Application: (Progress) Options: () Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77;tag=as01491743 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Type: application/sdp Content-Length: 258 v=0 o=root 1225456982 1225456982 IN IP4 176.249.0.77 s=Asterisk PBX 1.8.0 c=IN IP4 176.249.0.77 t=0 0 m=audio 15918 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- AGI Script Executing Application: (Playback) Options: (congestion,noanswer) -- SIP/1000-0019 Playing 'congestion.slin' (language 'en') -- SIP/1000-0019AGI Script agi.php completed, returning 0 Regards, Zohair Raza On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote: Hey Danny, I've this thing exactly running and working as Zohair mentioned! i.e I do not answer() the call rather put a progress() and soon after that playing back the sound file from playback with noanswer and then I get the file streaming as 183-Session progress file. I do understand that playing any sound file before establishing any audio session
Re: [asterisk-users] Playback with noanswer in AGI
Yes, Thanks Regards, Zohair Raza On Tue, Feb 7, 2012 at 1:37 PM, Sammy Govind govoi...@gmail.com wrote: Exactly that's what I expected. Great - now have fun On Tue, Feb 7, 2012 at 2:09 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: Sammy, Problem is at phones, with a linksys phone it works but with eyebeam and fanvill it doesn't Maybe they don't support early media. I think i will have to stick with ResetCDR and that will be okay now as I've modified the code for that Thank you Regards, Zohair Raza On Tue, Feb 7, 2012 at 12:09 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hi Sammy, Thanks for input. I have an eyebeam softphone registered with Asterisk 1.8.6 locally and from agi, I pass this $filetoplay = 'congestion'; $agi-exec(Progress); $agi-exec(Playback $filetoplay,noanswer); Have tried putting file in .gsm and .wav formats, I hear ringing tone instead of playback Please have a look at sip-trace --- SIP read from UDP:176.249.0.50:8721 --- INVITE sip:100@176.249.0.77 SIP/2.0 To: sip:100@176.249.0.77 From: Zohairsip:1000@176.249.0.77;tag=7f222672 Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport Call-ID: 2932f90ef302332b CSeq: 2 INVITE Contact: sip:1000@176.249.0.50:8721 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3006o stamp 17551 Authorization: Digest username=1000,realm=asterisk,nonce=2abce759,uri= sip:100@176.249.0.77 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5 Content-Length: 269 v=0 o=- 4333518 4333604 IN IP4 176.249.0.50 s=eyeBeam c=IN IP4 176.249.0.50 t=0 0 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 11 lines) --- Sending to 176.249.0.50:8721 (no NAT) sing INVITE request as basis request - 2932f90ef302332b Found peer '1000' for '1000' from 176.249.0.50:8721 == Using SIP RTP CoS mark 5 Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 101 Found audio description format speex for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 176.249.0.50:6506 Looking for 100 in default (domain 176.249.0.77) list_route: hop: sip:1000@176.249.0.50:8721 --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Length: 0 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID) -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php -- AGI Script Executing Application: (Progress) Options: () Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77;tag=as01491743 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Type: application/sdp Content-Length: 258 v=0 o=root 1225456982 1225456982 IN IP4 176.249.0.77 s=Asterisk PBX 1.8.0 c=IN IP4 176.249.0.77 t=0 0 m=audio 15918 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- AGI Script Executing Application: (Playback) Options: (congestion,noanswer) -- SIP/1000-0019 Playing 'congestion.slin' (language 'en') -- SIP/1000-0019AGI Script agi.php completed, returning 0 Regards, Zohair Raza On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.comwrote: Hey Danny, I've this thing exactly running and working as Zohair mentioned! i.e I do not answer
Re: [asterisk-users] Playback with noanswer in AGI
Confirmed as well, played back with wireshark and audio was there but phone was ringing. Thanks again. Regards, Zohair Raza On Tue, Feb 7, 2012 at 1:37 PM, Sammy Govind govoi...@gmail.com wrote: Hi, Given invites seems fine, can you take a wireshark trace of the call on your eyebeam machine! from that wireshark trace use telephony calls options and hear if you are actually receiving RTPs on your system. If you could hear the played back sound file on your eyembeam machine . this would mean that your eyebeam client is not good enough to play media while its in 183 session progress. Also can you send me the short sample php-agi script you are executing so i actually test this on my virtual machines as well. Regards, Sammy On Tue, Feb 7, 2012 at 1:09 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: Hi Sammy, Thanks for input. I have an eyebeam softphone registered with Asterisk 1.8.6 locally and from agi, I pass this $filetoplay = 'congestion'; $agi-exec(Progress); $agi-exec(Playback $filetoplay,noanswer); Have tried putting file in .gsm and .wav formats, I hear ringing tone instead of playback Please have a look at sip-trace --- SIP read from UDP:176.249.0.50:8721 --- INVITE sip:100@176.249.0.77 SIP/2.0 To: sip:100@176.249.0.77 From: Zohairsip:1000@176.249.0.77;tag=7f222672 Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport Call-ID: 2932f90ef302332b CSeq: 2 INVITE Contact: sip:1000@176.249.0.50:8721 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3006o stamp 17551 Authorization: Digest username=1000,realm=asterisk,nonce=2abce759,uri= sip:100@176.249.0.77 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5 Content-Length: 269 v=0 o=- 4333518 4333604 IN IP4 176.249.0.50 s=eyeBeam c=IN IP4 176.249.0.50 t=0 0 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 11 lines) --- Sending to 176.249.0.50:8721 (no NAT) sing INVITE request as basis request - 2932f90ef302332b Found peer '1000' for '1000' from 176.249.0.50:8721 == Using SIP RTP CoS mark 5 Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 101 Found audio description format speex for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 176.249.0.50:6506 Looking for 100 in default (domain 176.249.0.77) list_route: hop: sip:1000@176.249.0.50:8721 --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Length: 0 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID) -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php -- AGI Script Executing Application: (Progress) Options: () Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77;tag=as01491743 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Type: application/sdp Content-Length: 258 v=0 o=root 1225456982 1225456982 IN IP4 176.249.0.77 s=Asterisk PBX 1.8.0 c=IN IP4 176.249.0.77 t=0 0 m=audio 15918 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- AGI Script Executing Application: (Playback) Options: (congestion,noanswer) -- SIP/1000-0019 Playing 'congestion.slin' (language 'en') -- SIP/1000-0019AGI Script agi.php completed
Re: [asterisk-users] Asterisk V/s FreeSwitch
You mean concurrent calls? You can have several 100 concurrent calls with a good CPU in newer versions of asterisk, however calls per secons (CPS) have some limitations I guess reason being that both are different in Architecture, Asterisk was designed keeping PBX in mind but Freeswitch was for SIP switching Regards, Zohair Raza On Tue, Feb 7, 2012 at 3:38 PM, virendra bhati virbh...@gmail.com wrote: Hi List, 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... -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- 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] Asterisk V/s FreeSwitch
Virendra, You can test your box with sipp http://etel.wiki.oreilly.com/wiki/index.php/Using_SIPp_to_Stress_Test_Asterisk I have verified my Asterisk 1.8 box handling 500 concurrent calls and 15 calls per seconds with 20% cpu, without transcoding. Regards, Zohair Raza On Wed, Feb 8, 2012 at 11:53 AM, Brynjolfur Thorvardsson bi...@itanet.nuwrote: My Asterisk 1.4.19 happily excepts up to 80 concurrent calls (the most I’ve seen so far) which sends the CPU load up to ~20% on a fairly old server. In our busiest period, from 8 to 8:05 I see up to 200 incoming calls, somewhat less than one call/second. ** ** My superiors want to expand and increase the number of clients significantly and the scalability of Asterisk is beginning to worry me. Someone mentioned a “roof” of 250 CC in Asterisk after which stability and call quality becomes increasingly affected. ** ** My plan is to implement load-balancing using DUNDi with one extra server initially, and a second available on site for further expansion. This should enable me to accommodate ten times our current load without any significant problems (I hope!), and adding more servers is fairly easy (although I guess there are diminishing returns?). ** ** When it comes to the long term I must admit I am increasingly looking at trying out FreeSwitch, the configuration might be trickier but scalability is much higher on my list of priorities. ** ** *Fra:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *På vegne af *virendra bhati *Sendt:* 7. februar 2012 12:38 *Til:* Asterisk Users Mailing List - Non-Commercial Discussion *Emne:* [asterisk-users] Asterisk V/s FreeSwitch ** ** Hi List, 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... -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 ** ** -- _ -- 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] Asterisk V/s FreeSwitch
It's 4 core Intel(R) Xeon(R) CPUX3220 with 6GB RAM Regards, Zohair Raza On Wed, Feb 8, 2012 at 5:46 PM, Bryant Zimmerman brya...@zktech.com wrote: Zohair What kind of hardware spec are you running CPU, MEM, Drives? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- *From*: Zohair Raza engineerzuhairr...@gmail.com *Sent*: Wednesday, February 08, 2012 3:08 AM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] Asterisk V/s FreeSwitch Virendra, You can test your box with sipp http://etel.wiki.oreilly.com/wiki/index.php/Using_SIPp_to_Stress_Test_Asterisk I have verified my Asterisk 1.8 box handling 500 concurrent calls and 15 calls per seconds with 20% cpu, without transcoding. Regards, Zohair Raza On Wed, Feb 8, 2012 at 11:53 AM, Brynjolfur Thorvardsson bi...@itanet.nuwrote: My Asterisk 1.4.19 happily excepts up to 80 concurrent calls (the most I’ve seen so far) which sends the CPU load up to ~20% on a fairly old server. In our busiest period, from 8 to 8:05 I see up to 200 incoming calls, somewhat less than one call/second. My superiors want to expand and increase the number of clients significantly and the scalability of Asterisk is beginning to worry me. Someone mentioned a “roof” of 250 CC in Asterisk after which stability and call quality becomes increasingly affected. My plan is to implement load-balancing using DUNDi with one extra server initially, and a second available on site for further expansion. This should enable me to accommodate ten times our current load without any significant problems (I hope!), and adding more servers is fairly easy (although I guess there are diminishing returns?). When it comes to the long term I must admit I am increasingly looking at trying out FreeSwitch, the configuration might be trickier but scalability is much higher on my list of priorities. *Fra:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *På vegne af *virendra bhati *Sendt:* 7. februar 2012 12:38 *Til:* Asterisk Users Mailing List - Non-Commercial Discussion *Emne:* [asterisk-users] Asterisk V/s FreeSwitch Hi List, 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... -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- 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 -- _ -- 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] Is there any way to make call fail after # of rings?
Try this
exten= yournumberhere,1,Dial(SIP/peern1,60)
exten= yournumberhere,n,GotoIf($[${DIALSTATUS} != ANSWER]?4)
exten= yournumberhere,n,Hangup
exten= yournumberhere,n,Dial(SIP/peer2,60)
exten= yournumberhere,n,GotoIf($[${DIALSTATUS} != ANSWER]?9)
exten= yournumberhere,n,Hangup
you can add more conditions in the same way
Regards,
Zohair Raza
On Fri, Feb 17, 2012 at 1:00 PM, CDR vene...@gmail.com wrote:
My customer needs to set a forwarding based on number of rings,i.e.,
if the phone rings 5 times (user-selectable), then try another number.
Is there a way to do such a thing with Asterisk? I could not find way
to do it based on the documentation of the Dial function. The protocol
is SIP only, however, I could use a different one if it provided a
workaround. If this is the wrong tool for the job, what technology
would do this?
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Re: [asterisk-users] how many UDP ports is required for 1 call
Hi Kevin, http://www.freepbx.org/support/documentation/howtos/howto-setup-a-remote-sip-extension this says 4 active ports for one call Regards, Zohair Raza On Wed, Feb 22, 2012 at 4:38 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/22/2012 06:26 AM, virendra bhati wrote: Does anyone know the correct information of my question. All are move round and round . What does that mean? I answered your question with the correct and complete information. On Tue, Feb 21, 2012 at 7:28 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: On 02/21/2012 07:51 AM, Alex Balashov wrote: As many ports as required by the nature of the call, i.e. the protocol(s) used for the bearer. For an IAX2 call, the answer is 'zero' for all of those call types (at least the ones that are supported in IAX2, not all of them are). For protocols that use RTP for media transport, two ports are required for each media stream (one for RTP, one for RTCP). -- 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-**usershttp://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] AGI: blocking script until playback complete
Try passing escape character GET DATA $filename $timeout $max_digits $escape_character Regards, Zohair Raza On Wed, Feb 22, 2012 at 6:40 PM, Chris Bagnall aster...@lists.minotaur.ccwrote: Greetings list, I've done AGI scripting before, but in the past I've always wanted control to be returned to the dialplan as soon as possible. However, today I have a scenario where I want the script to remain running during the playback of a file so that I can read DTMF at the end of playback. However, doing this: GET DATA en_welcome 5000 6 Results (correctly) in the following in the asterisk console: -- SIP/a.b.c.d-dc027b50 Playing 'en_welcome' (language 'en') But the AGI continues to run on after this point, not waiting for either the sound file to be played, nor for the expected 6 DTMF digits. Adding a simple 10 second sleep/wait to the AGI allows the sound file to be successfully played back. I'm sure I must be missing something very obvious, buy my google-fu is failing me this afternoon. Suggestions gratefully received :-) Thanks in advance. Kind regards, Chris -- This email is made from 100% recycled electrons -- __**__**_ -- 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-**usershttp://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] AGI: blocking script until playback complete
I gave it from phpagi. It works for me using phpagi's function get_data http://phpagi.sourceforge.net/phpagi22/api-docs/phpAGI/AGI.html Regards, Zohair Raza On Wed, Feb 22, 2012 at 7:20 PM, Chris Bagnall aster...@lists.minotaur.ccwrote: The problem seems to be that GET DATA returns control to the script before the audio file has even played, let alone any DTMF tones have been entered. I would have expected script execution to be blocked until the result from GET DATA was available. -- _ -- 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] 10.2.0-rc2: permitted contact can't register.
You want to allow single IP or whole subnet ? Regards, Zohair Raza On Wed, Feb 29, 2012 at 4:44 AM, sean darcy seandar...@gmail.com wrote: An outside device can't register: WARNING: getnameinfo(): ai_family not supported WARNING: chan_sip.c:14456 parse_register_contact: Domain '69.xxx.yyy.zzz:5060' disallowed by contact ACL (violating IP ) sip.conf: [general] ... alwaysreject=yes dynamic_exclude_static = yes allowguest=no contactdeny=0.0.0.0/0.0.0.0 contactpermit=69.0.0.0/255.0.**0.0 http://69.0.0.0/255.0.0.0 I've also tried without any contactdeny. Same result. I'm completely puzzled. Any help appreciated. sean -- __**__**_ -- 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-**usershttp://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] Ongoing attack from 188.138.100.16
Hi, this can also be helpful http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood/ Regards, Zohair Raza On Wed, Mar 7, 2012 at 7:53 PM, Danny Nicholas da...@debsinc.com wrote: Nothing against fail2ban but in this case I think the “route drop” solution is more appropriate. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Wednesday, March 07, 2012 9:52 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Ongoing attack from 188.138.100.16 ** ** fail2ban works perfect!! On Wed, Mar 7, 2012 at 12:47 PM, Jamie A. Stapleton jstaple...@computer-business.com wrote: Block them. They are one of the Internet's top bad IP addresses. http://www.threatstop.com/checkip -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Tuesday, March 06, 2012 7:29 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Ongoing attack from 188.138.100.16 I've been logging sip registrations from this IP address for 2 days now. I've emailed the domain's admin, but nothing seems to come of it. I've routed him into oblivion, but still, I think 50 requests a second for 2 days is a bit much. Any ideas? -- Take care and have fun, Mike Diehl. -- _ -- 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 ** ** -- _ -- 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 insecure
Hi, How to allow registered sip users to call without re-authentication insecure =yes/very are deprecated in 1.8 I want to avoid fromuser= in peer configuration. When I add this in peer asterisk, my asterisk accepts call otherwise it says username mismatch. Please help Regards, Zohair Raza -- _ -- 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 insecure
They don't require authentication of invites which I do need Regards, Zohair Raza On Thu, Mar 22, 2012 at 3:20 PM, Leandro Dardini ldard...@gmail.com wrote: 2012/3/22 Zohair Raza engineerzuhairr...@gmail.com Hi, How to allow registered sip users to call without re-authentication insecure =yes/very are deprecated in 1.8 I want to avoid fromuser= in peer configuration. When I add this in peer asterisk, my asterisk accepts call otherwise it says username mismatch. Please help Regards, Zohair Raza There are other options, like invite and port to be used when you trust the IP of the caller. Leandro -- _ -- 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] Sip insecure
My main box is asterisk 1.8 and there are two boxes, one asterisk 1.8 and other 1.4 with 1.4, I don't need to define fromuser=username but in 1.8 I can't make call without it the problem in defining fromuser is, it overrides the callerid Main box has same settings for both peers Regards, Zohair Raza On Thu, Mar 22, 2012 at 3:26 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: They don't require authentication of invites which I do need Regards, Zohair Raza On Thu, Mar 22, 2012 at 3:20 PM, Leandro Dardini ldard...@gmail.comwrote: 2012/3/22 Zohair Raza engineerzuhairr...@gmail.com Hi, How to allow registered sip users to call without re-authentication insecure =yes/very are deprecated in 1.8 I want to avoid fromuser= in peer configuration. When I add this in peer asterisk, my asterisk accepts call otherwise it says username mismatch. Please help Regards, Zohair Raza There are other options, like invite and port to be used when you trust the IP of the caller. Leandro -- _ -- 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] Sip insecure
I've figured this out using match_auth_username =yes Thanks Regards, Zohair Raza On Thu, Mar 22, 2012 at 3:33 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: My main box is asterisk 1.8 and there are two boxes, one asterisk 1.8 and other 1.4 with 1.4, I don't need to define fromuser=username but in 1.8 I can't make call without it the problem in defining fromuser is, it overrides the callerid Main box has same settings for both peers Regards, Zohair Raza On Thu, Mar 22, 2012 at 3:26 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: They don't require authentication of invites which I do need Regards, Zohair Raza On Thu, Mar 22, 2012 at 3:20 PM, Leandro Dardini ldard...@gmail.comwrote: 2012/3/22 Zohair Raza engineerzuhairr...@gmail.com Hi, How to allow registered sip users to call without re-authentication insecure =yes/very are deprecated in 1.8 I want to avoid fromuser= in peer configuration. When I add this in peer asterisk, my asterisk accepts call otherwise it says username mismatch. Please help Regards, Zohair Raza There are other options, like invite and port to be used when you trust the IP of the caller. Leandro -- _ -- 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] Video Conference in Asterisk1.4 (using asterisk gui)
videosupport=yes in sip.conf Regards, Zohair Raza On Mon, Apr 9, 2012 at 12:22 PM, p070075 Muhammad Atif Ramzan p070...@nu.edu.pk wrote: Hi I am new to asterisk 1.4 can someone tell about how to enable the video conference in asterisk-gui 2.0. -- _ -- 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] Asterisk 1.8.10 getaddrinfo
your destination address is not correct, on CLI, check what is actually being passed in Dial application Regards, Zohair Raza On Wed, Apr 18, 2012 at 2:04 AM, motty.cruz motty.c...@gmail.com wrote: Hello All, I'm gettint this error, started recently when I upgraded to 1.8.10 from 1.8.4. [Apr 17 08:03:52] ERROR[9099]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo(external out, (null), ...): Name or service not known [Apr 17 08:03:52] WARNING[9099]: chan_sip.c:26503 sip_request_call: Unable to find IP address for host externalout. We will not use this remote IP address Does anybody have an idea how to fix error above? Thanks, motty -- _ -- 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] Setting channel variable using AMI
Hi list, I am trying to set a channel variable from AMI, when I do so I get success response but there is no variable set to that channel. http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+SetVar When I don't pass channel name for setting a global variable, I can get that variable in hangup extension but not the channel variable. Regards, Zohair Raza -- _ -- 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] Need queue name in CDR
Hi, http://www.voip-info.org/wiki/view/Asterisk+log+queue_log http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL Regards, Zohair Raza On Wed, Jun 13, 2012 at 7:38 AM, Pratik Shrestha pratik...@gmail.com wrote: Dear All, I am making asterisk report using CDR values given by asterisk. I have queues which consist of multiple members (extension). Also, an extension may be in multiple queues. So, I want CDR to record the name/number of queue from which the call was originated. E.g. Channel DestinationChannel Src Destination SIP/KOT-000c Local/102@from-queue-6a84;1 0856511524 (first line in CDR) Local/102@from-queue-6a84;2 SIP/102-000e 0856511524 102 (second line in CDR) In above example, is a queue and 102 is an extension which is member to that queue. So call comes from 0856511524 and goes to queue first and queue routes call to 102 extension. So what I need is when the queue is routed to extension 102 (in the seconds line), I want to show the queue () also. I know that I can track the queue by comparing Destination Channel of queue(first line) with Channel of extension (second line). But this will make my query very long and hard. Please help me. I am still new to asterisk. Regards, Pratik -- _ -- 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] Does Asterisk support AMR and AMR-WB
I patched asterisk once 2 years ago but couldn't exactly remember the way or link may be one of these that google showed me today. you can give a try http://www.howtonix.com/amr-codec-for-asterisk-1-4-and-1-6/ http://sourceforge.net/projects/asterisk-amr/ It worked for me, by chance I have cli output in my mail which I sent to my manager after installing the codec. sandbox*CLI core show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 amr g723- ---- -- -- -- -- - gsm- -222 21 3- -- 2- 14 ulaw- 2-12 21 3- -- 2- 14 alaw- 21-2 21 3- -- 2- 14 g726aal2- 222- 21 3- -- 1- 14 adpcm- 2222 -1 3- -- 2- 14 slin- 1111 1- 2- -- 1- 13 lpc10- 2222 21 -- -- 2- 14 g729- ---- -- -- -- -- - speex- ---- -- -- -- -- - ilbc- ---- -- -- -- -- - g726- 2221 21 3- -- -- 14 g722- ---- -- -- -- -- - amr- 2222 21 3- -- 2- - Regards, Zohair Raza www.zuhair.info http://ae.linkedin.com/in/zuhairraza On Fri, Jun 15, 2012 at 7:52 AM, Jakson Kalsson sipmaill...@gmail.com wrote: Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the Asterisk support these codecs and RFC4867 ? If no, there has any plugin to support this ? Also, any other Server/PBX which support AMR, AMR-WB recommended are welcome. Best 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
[asterisk-users] Strange behavior - Can't figure out
Hi, I have two asterisk boxes, one with asterisk 1.8.12.0 and the other with asterisk 1.8.9.2 Sip show settings of both boxes have no difference and also the peers I am generating a call using call file with following details: Channel: SIP/1028 Account: 9164421122 -- this is the accountcode of 1028 MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: default Extension: 1031 Priority: 1 CallerID: Zohair Raza1031 -- I want to see this caller id at dialing peer (1028) and Test 1028 (originiating caller id) at dialed peer On asterisk 1.8.9.2 I get results as expected and debug output is as below -- Executing [1031@default:1] AGI(SIP/1028-3897, agi.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php SIP/1028-3897AGI Tx agi_request: agi.php SIP/1028-3897AGI Tx agi_channel: SIP/1028-3897 SIP/1028-3897AGI Tx agi_language: en SIP/1028-3897AGI Tx agi_type: SIP SIP/1028-3897AGI Tx agi_uniqueid: a-1340263981.14503 SIP/1028-3897AGI Tx agi_version: 1.8.9.2 SIP/1028-3897AGI Tx agi_callerid: 1028 SIP/1028-3897AGI Tx agi_calleridname: Test -- caller id of 1028 SIP/1028-3897AGI Tx agi_callingpres: 0 SIP/1028-3897AGI Tx agi_callingani2: 0 SIP/1028-3897AGI Tx agi_callington: 0 SIP/1028-3897AGI Tx agi_callingtns: 0 SIP/1028-3897AGI Tx agi_dnid: unknown SIP/1028-3897AGI Tx agi_rdnis: unknown SIP/1028-3897AGI Tx agi_context: default SIP/1028-3897AGI Tx agi_extension: 1031 SIP/1028-3897AGI Tx agi_priority: 1 SIP/1028-3897AGI Tx agi_enhanced: 0.0 SIP/1028-3897AGI Tx agi_accountcode: 9164421122 -- accountcode of 1028 here SIP/1028-3897AGI Tx agi_threadid: 1095772480 SIP/1028-3897AGI Tx SIP/1028-3897AGI Rx GET VARIABLE CDR(clid) SIP/1028-3897AGI Tx 200 result=1 (Test 1028) Same I am trying on another box with these details Channel: SIP/5405 Account: 6167531316 -- this is the accountcode of 5405 MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: default Extension: 5050 Priority: 1 CallerID: Test 25050 -- I want to see this caller id at dialing peer (5405) and Test 5050 (originiating caller id) at dialed peer But, for some reason it is showing Test 2 5050 on both phones. On Cli Debug, the behavior is also different -- Executing [5050@default:1] AGI(SIP/5405-01f7, agi.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php SIP/5405-01f7AGI Tx agi_request: agi.php SIP/5405-01f7AGI Tx agi_channel: SIP/5405-01f7 SIP/5405-01f7AGI Tx agi_language: en SIP/5405-01f7AGI Tx agi_type: SIP SIP/5405-01f7AGI Tx agi_uniqueid: TT-1340270088.522 SIP/5405-01f7AGI Tx agi_version: 1.8.12.0 SIP/5405-01f7AGI Tx agi_callerid: 5050 SIP/5405-01f7AGI Tx agi_calleridname: Test 2 -- here it's callerid of 5050 instead of 5405 SIP/5405-01f7AGI Tx agi_callingpres: 0 SIP/5405-01f7AGI Tx agi_callingani2: 0 SIP/5405-01f7AGI Tx agi_callington: 0 SIP/5405-01f7AGI Tx agi_callingtns: 0 SIP/5405-01f7AGI Tx agi_dnid: unknown SIP/5405-01f7AGI Tx agi_rdnis: unknown SIP/5405-01f7AGI Tx agi_context: default SIP/5405-01f7AGI Tx agi_extension: 5050 SIP/5405-01f7AGI Tx agi_priority: 1 SIP/5405-01f7AGI Tx agi_enhanced: 0.0 SIP/5405-01f7AGI Tx agi_accountcode: 6167531316 -- account code of 5405 SIP/5405-01f7AGI Tx agi_threadid: 1084270912 SIP/5405-01f7AGI Tx SIP/5405-01f7AGI Rx GET VARIABLE CDR(clid) SIP/5405-01f7AGI Tx 200 result=1 (Test 2 5050) Can anybody help me on figuring this out please. Thanks Regards, Zohair Raza -- _ -- 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] How to set SIP to auto answer in the dial plan .
try with SipAddHeader(uri=answer-after=0) check syntax for Addheader Regards, Zohair Raza On Fri, Jul 13, 2012 at 1:42 PM, upendra uppi...@gmail.com wrote: Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -- _ -- 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] How to set SIP to auto answer in the dial plan .
In dialplan http://www.voip-info.org/wiki/view/Asterisk+cmd+SipAddHeader Regards, Zohair Raza On Fri, Jul 13, 2012 at 1:50 PM, upendra uppi...@gmail.com wrote: Hi, thanks , i need to put this in the sip context... regards Upendra. On Fri, Jul 13, 2012 at 3:15 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: try with SipAddHeader(uri=answer-after=0) check syntax for Addheader Regards, Zohair Raza On Fri, Jul 13, 2012 at 1:42 PM, upendra uppi...@gmail.com wrote: Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -- _ -- 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 -- _ -- 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] multiple users for jabber.conf
Also you could have a look at openfire and it's Asterisk-IM plugin On Wed, Sep 12, 2012 at 10:41 AM, Hans Witvliet aster...@a-domani.nlwrote: 1.8 machine -- _ -- 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] Allowing peers from specific subnet only
Hi You can achieve this with either permit/deny or contactpermit/contactdeny Single IP should be defined like : deny=0.0.0.0/0.0.0.0 permit=192.168.2.1/255.255.255.255 And networks in similar way with appropriate subnet mask deny=0.0.0.0/0.0.0.0 permit=192.168.2.0/255.255.255.0 You can also specify multiple subnets with ';' like: permit=192.168.2.0/255.255.255.0;192.168.1.0/255.255.255.0 Regards, Zohair Raza On Mon, Nov 19, 2012 at 4:12 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi; How I can make my configuration to allow the sip phones only from specific IP addresses range (for example from 192.168.10.1 - 192.168.10.50) to be allowed to connect for asterisk? In other words, in addition to be authenticated based on the username and password, it is required that the IP address of the Phone to be from this range. How? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] block one number in incoming calls
exten =520xx,1,NoOp(Caller-ID: ${CALLERID(all)})
exten =520xx,2,GotoIf($[${CALLERID(num)} = 0666XX ]?3:4)
exten = 520xx,3,Dial(SIP/224, 30)
exten = 520xx,4,hangup
Regards,
Zohair Raza
On Mon, Jan 14, 2013 at 7:43 PM, Michelle Dupuis mdup...@ocg.ca wrote:
Check out smartCID on www.generationd.com
This script allows lookup of incomming calls based on number and either
Block (no ring), endless ring (ignore), or pass through to asterisk. It
allows allows rewriting of CID name based on number. All numbers stored in
a mysql table. A free script.
It also does reverse look of CID based on number using a variety of free
web sites (but that's intended for experimentation only)
--
*From:* asterisk-users-boun...@lists.digium.com [
asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit
[salah.elharit...@gmail.com]
*Sent:* Monday, January 14, 2013 10:33 AM
*To:* Asterisk Users List
*Subject:* [asterisk-users] block one number in incoming calls
Hello list
could you please help me about one question.
i have asterisk 1.4 installed, i configure the inbound call in my
asterisk like below.
exten = 520xx,1,Dial(SIP/224, 30).
when the customer call my number (520xx) the sip phone 224 works
without issue
my problem i have a lot of calls coming from this number (0666xx) and
i want to block it.
if you can give me an example please .
thanks and regards
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Re: [asterisk-users] AGI command
you need to run full command, like agi show commands topic answer agi show commands topic gosub agi set debug on Regards, Zohair Raza On Tue, Jan 15, 2013 at 1:05 PM, Muhammad mohammad.ghaz...@gmail.comwrote: Hi, in CLI, I type agi show or other agi commad, but response me command not found. How can see agi is work normally in my server? -- _ -- 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] AGI command
On Wed, Jan 16, 2013 at 11:01 AM, Muhammad mohammad.ghaz...@gmail.comwrote: *Thanks Zohair! I wrote some php code to working with AGI, but it dosen't work. * *I don't know how can run it. please explain me when I put my php code inside /var/lib/asterisk/agi-bin so, what should I do after that. * Make sure Asterisk has access to your AGI script, and make it executable (chmod u+x agi.php). Also make sure it has shebang (!#/usr/bin/php) *and the second one, how can limit users to call just my number in list at database and permit to call another numbers.* * * That depends on logic in your script, you can also separate users by contexts * * On Tue, Jan 15, 2013 at 12:39 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: you need to run full command, like agi show commands topic answer agi show commands topic gosub agi set debug on Regards, Zohair Raza On Tue, Jan 15, 2013 at 1:05 PM, Muhammad mohammad.ghaz...@gmail.comwrote: Hi, in CLI, I type agi show or other agi commad, but response me command not found. How can see agi is work normally in my server? -- _ -- 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 -- _ -- 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] Cisco 7942 Connected line ID
Hi, Is it working for anyone? I have tried with trustrpid=yes sendrpid=yes/pai but can not get it working, Asterisk cli shows prevented message like this. Connected line update to SIP/1231-0200 prevented Regards, Zohair Raza -- _ -- 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] Cisco 7942 Connected line ID
Thanks for pointing that have it disabled now But caller id still not getting updated -- _ -- 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] Cisco 7942 Connected line ID
It works fine on my SPA504G but not on 7942 Regards, Zohair Raza On Sat, Feb 16, 2013 at 9:32 AM, Vladimir Mikhelson v...@mikhelson.comwrote: Zohair, I am not sure about the specifics of 7942 as I use 7906. Connected line CID shows up on my 7906 with the following sip.conf settings: - trustrpid=yes - sendrpid=yes -Vladimir On 2/15/2013 11:09 AM, Zohair Raza wrote: Hi, Is it working for anyone? I have tried with trustrpid=yes sendrpid=yes/pai but can not get it working, Asterisk cli shows prevented message like this. Connected line update to SIP/1231-0200 prevented Regards, Zohair Raza -- _ -- 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 -- _ -- 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] Asterisk crashed
Hi, I am running asterisk 1.8.14.0, It was running fine for last few days and suddenly crashed today In logs I can see that abrt tried to save the core dump but it couldn't Mar 6 12:11:09 localhost kernel: asterisk[26544]: segfault at 72656d69ac ip 00533c19 sp 7f7db9ce3af0 error 4 in asterisk[40+1d1000] Mar 6 12:11:15 localhost abrt[31287]: Saved core dump of pid 26528 (/usr/sbin/asterisk) to /var/spool/abrt/ccpp-2013-03-06-12:11:09-26528 (450703360 bytes) Mar 6 12:11:15 localhost abrtd: Directory 'ccpp-2013-03-06-12:11:09-26528' creation detected Mar 6 12:11:15 localhost abrtd: Executable '/usr/sbin/asterisk' doesn't belong to any package Mar 6 12:11:15 localhost abrtd: 'post-create' on '/var/spool/abrt/ccpp-2013-03-06-12:11:09-26528' exited with 1 *Asterisk was running as root user Any suggestions? Regards, Zohair Raza -- _ -- 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] Asterisk crashed
Thank you both Matthew, I can not do that because core file is not available Bharat is right, the file was not written because of abrt's issue https://bugzilla.redhat.com/show_bug.cgi?id=768149 I am turning it off now, I hope asterisk won't crash again but in any case if it does I will have a core dump because it is started with safe_asterisk Thanks again Regards, Zohair Raza On Thu, Mar 7, 2013 at 10:52 PM, Bharat Lalcheta bharatlalch...@gmail.comwrote: Did u test it without abrt? On Mar 7, 2013 10:03 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: Its Centos 6 with kernel 2.6.32-279.19.1.el6.x86_64 Regards, Zohair Raza On Thu, Mar 7, 2013 at 8:28 AM, Bharat Lalcheta bharatlalch...@gmail.com wrote: Can you provide OS details ? Its seems problem of abrt. Did u tested asterisk without abrt Regards, Bharat Lalcheta On Thu, Mar 7, 2013 at 12:05 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hi, I am running asterisk 1.8.14.0, It was running fine for last few days and suddenly crashed today In logs I can see that abrt tried to save the core dump but it couldn't Mar 6 12:11:09 localhost kernel: asterisk[26544]: segfault at 72656d69ac ip 00533c19 sp 7f7db9ce3af0 error 4 in asterisk[40+1d1000] Mar 6 12:11:15 localhost abrt[31287]: Saved core dump of pid 26528 (/usr/sbin/asterisk) to /var/spool/abrt/ccpp-2013-03-06-12:11:09-26528 (450703360 bytes) Mar 6 12:11:15 localhost abrtd: Directory 'ccpp-2013-03-06-12 :11:09-26528' creation detected Mar 6 12:11:15 localhost abrtd: Executable '/usr/sbin/asterisk' doesn't belong to any package Mar 6 12:11:15 localhost abrtd: 'post-create' on '/var/spool/abrt/ccpp-2013-03-06-12:11:09-26528' exited with 1 *Asterisk was running as root user Any suggestions? Regards, Zohair Raza -- _ -- 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 -- Bharat Lalcheta -- _ -- 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 -- _ -- 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] Cisco 7942 Connected line ID
Just to add that it was fixed by using this patch https://issues.asterisk.org/jira/browse/ASTERISK-13145 It also made cisco softkeys working and call transfer/3 way conference as well Regards, Zohair Raza On Sun, Feb 17, 2013 at 2:58 AM, Vladimir Mikhelson v...@mikhelson.comwrote: Zohair, SPA504G is LinkSys. It is completely different. 7942 and 7906 are true Cisco phones. That was why I gave you this example. -Vladimir On 2/16/2013 6:04 AM, Zohair Raza wrote: It works fine on my SPA504G but not on 7942 Regards, Zohair Raza On Sat, Feb 16, 2013 at 9:32 AM, Vladimir Mikhelson v...@mikhelson.comwrote: Zohair, I am not sure about the specifics of 7942 as I use 7906. Connected line CID shows up on my 7906 with the following sip.conf settings: - trustrpid=yes - sendrpid=yes -Vladimir On 2/15/2013 11:09 AM, Zohair Raza wrote: Hi, Is it working for anyone? I have tried with trustrpid=yes sendrpid=yes/pai but can not get it working, Asterisk cli shows prevented message like this. Connected line update to SIP/1231-0200 prevented Regards, Zohair Raza _ -- 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 -- _ -- 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] Asterisk SIP TCP
Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to 10.200.1.55:5076 returned -2: Interrupted syste Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message [2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to 10.200.1.55:5075: Connection refused [2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded. I will produce a back trace later today and file a bug, I am using version 1.8.14.0 Please note, I have to stick with TCP because of packet loss in the network Any suggestions? Regards, Zohair Raza -- _ -- 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] Asterisk SIP TCP
Backtrace and logs attached here : https://issues.asterisk.org/jira/browse/ASTERISK-21447 Regards, Zohair Raza On Mon, Apr 15, 2013 at 11:13 PM, Mark Henry markhenry...@gmail.com wrote: this is my secondary email Regards Zohair On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry markhenry...@gmail.comwrote: Tried disabling qualify and changing frequency with qualify=yes already, no luck :( On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf mehroz.ashra...@gmail.com wrote: I believe qualify parameters does help in doing so. Asterisk forgets about the peer info when qualify are not acknowledged. You can also check qualifyfreq to limit the number of qualifies for particular peer. On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to 10.200.1.55:5076 returned -2: Interrupted syste Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message [2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to 10.200.1.55:5075: Connection refused [2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded. I will produce a back trace later today and file a bug, I am using version 1.8.14.0 Please note, I have to stick with TCP because of packet loss in the network Any suggestions? Regards, Zohair Raza -- _ -- 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 -- _ -- 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] Asterisk SIP TCP
Here is what I have, also attached sip show settings output and part of sip.conf in issues [general] udpbindaddr=172.20.255.40 transport=udp,tcp tcpenable=yes tlsenable=no tcpbindaddr=172.20.255.40 directrtpsetup=no directmedia=yes allowguest=no match_auth_username=yes tos_sip=AF31 tos_audio=ef tos=0xB8 tos_video=af41 ; Sets TOS for RTP video packets. tos_text=af41 ; Sets TOS for RTP text packets. trustrpid = yes ; If Remote-Party-ID should be trusted sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no) disallow=all allow=alaw allow=ulaw allow=g729 maxforwards=70 relaxdtmf=yes rpid_update = yes maxexpiry=400 minexpiry=60 defaultexpiry=300 qualify=yes ; notifycid = yes ; Control whether caller ID information is sent along with dialog-info+xml notifications (supported by snom phones) qualifyfreq=300 qualifypeers=1 qualifygap=2000 registertimeout=20 registerattempts=10 progressinband=never ignoreregexpire=yes On Tue, Apr 16, 2013 at 9:44 AM, Bharat Lalcheta bharatlalch...@gmail.comwrote: Can you give sip.conf ? I am using asterisk 1.8.15 on both udp and tcp and not able to generate this scenario. Regards, Bharat Lalcheta On Tue, Apr 16, 2013 at 11:03 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Backtrace and logs attached here : https://issues.asterisk.org/jira/browse/ASTERISK-21447 Regards, Zohair Raza On Mon, Apr 15, 2013 at 11:13 PM, Mark Henry markhenry...@gmail.comwrote: this is my secondary email Regards Zohair On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry markhenry...@gmail.comwrote: Tried disabling qualify and changing frequency with qualify=yes already, no luck :( On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf mehroz.ashra...@gmail.com wrote: I believe qualify parameters does help in doing so. Asterisk forgets about the peer info when qualify are not acknowledged. You can also check qualifyfreq to limit the number of qualifies for particular peer. On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to 10.200.1.55:5076 returned -2: Interrupted syste Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message [2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to 10.200.1.55:5075: Connection refused [2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded. I will produce a back trace later today and file a bug, I am using version 1.8.14.0 Please note, I have to stick with TCP because of packet loss in the network Any suggestions? Regards, Zohair Raza -- _ -- 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 -- _ -- 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 -- Bharat Lalcheta
Re: [asterisk-users] Asterisk SIP TCP
On Tue, Apr 16, 2013 at 10:12 AM, Bharat Lalcheta bharatlalch...@gmail.comwrote: ;ignoreregexpire=yes; Enabling this setting has two functions: ; ; For non-realtime peers, when their registration expires, the ; information will _not_ be removed from memory or the Asterisk database ; if you attempt to place a call to the peer, the existing information ; will be used in spite of it having expired ; ; For realtime peers, when the peer is retrieved from realtime storage, ; the registration information will be used regardless of whether ; it has expired or not; if it expires while the realtime peer ; is still in memory (due to caching or other reasons), the ; information will not be removed from realtime storage I tried setting it to no already, but asterisk was keep trying to establish connection at old ip and port Also remove all qualify related parameters and keepalive if set when qualify is set to no, does qualifyfreq have an effect? because I tried qualify=no bu the qualifyfreq was set at that time, I set qualifyfreq=300 but requests were going every few seconds (around 30 secs) One thing I doubt is Insecure field, it is set to no at the moment. By name it is for security only but setting it insecure=port may effect? Hope it will solve your problem Regards, Bharat Lalcheta On Tue, Apr 16, 2013 at 11:26 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Here is what I have, also attached sip show settings output and part of sip.conf in issues [general] udpbindaddr=172.20.255.40 transport=udp,tcp tcpenable=yes tlsenable=no tcpbindaddr=172.20.255.40 directrtpsetup=no directmedia=yes allowguest=no match_auth_username=yes tos_sip=AF31 tos_audio=ef tos=0xB8 tos_video=af41 ; Sets TOS for RTP video packets. tos_text=af41 ; Sets TOS for RTP text packets. trustrpid = yes ; If Remote-Party-ID should be trusted sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no) disallow=all allow=alaw allow=ulaw allow=g729 maxforwards=70 relaxdtmf=yes rpid_update = yes maxexpiry=400 minexpiry=60 defaultexpiry=300 qualify=yes ; notifycid = yes ; Control whether caller ID information is sent along with dialog-info+xml notifications (supported by snom phones) qualifyfreq=300 qualifypeers=1 qualifygap=2000 registertimeout=20 registerattempts=10 progressinband=never ignoreregexpire=yes On Tue, Apr 16, 2013 at 9:44 AM, Bharat Lalcheta bharatlalch...@gmail.com wrote: Can you give sip.conf ? I am using asterisk 1.8.15 on both udp and tcp and not able to generate this scenario. Regards, Bharat Lalcheta On Tue, Apr 16, 2013 at 11:03 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Backtrace and logs attached here : https://issues.asterisk.org/jira/browse/ASTERISK-21447 Regards, Zohair Raza On Mon, Apr 15, 2013 at 11:13 PM, Mark Henry markhenry...@gmail.comwrote: this is my secondary email Regards Zohair On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry markhenry...@gmail.comwrote: Tried disabling qualify and changing frequency with qualify=yes already, no luck :( On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf mehroz.ashra...@gmail.com wrote: I believe qualify parameters does help in doing so. Asterisk forgets about the peer info when qualify are not acknowledged. You can also check qualifyfreq to limit the number of qualifies for particular peer. On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609
Re: [asterisk-users] Is Asternic.net out of business (Flash Operator, Call Center Stats)?
My experience was good, Nicolas was very helpful and quick Regards, Zohair Raza On Tue, Jun 18, 2013 at 4:26 AM, Carlos Alvarez car...@televolve.comwrote: No vacation notice, nothing, other than the system auto-replying saying that the ticket will be closed because we didn't have any action on it. Rather distressing for our customers. On Mon, Jun 17, 2013 at 5:22 PM, Gregory Malsack gmals...@coastalacq.comwrote: No. Although Nicolas may have gone on holiday. I just purchased 2 licenses for fop2 a month or so ago. Carlos Alvarez car...@televolve.com wrote: We have licensed both products and sent a support request on 6/11, with zero reply or any activity on it at all so far. No replies to subsequent ticket updates or e-mails. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Asterisk listening on undefined IP as per bindaddr
Hello all, I am running asterisk on VMs with standby heartbeat configuration, Heartbeat assigns a virtual IP 172.20.255.40 on machine afterwards asterisk is started. In the sip.conf, I have explicitly define bindaddr=172.20.255.40 but sometimes I see packets coming from physical IP 172.20.255.41 I have both tcp and udp transport enabled Here is the lsof -ni :5060 output asterisk 2878 asterisk 613r IPv4 40060683 0t0 TCP 172.20.255.41:52381-10.100.210.110:sip (ESTABLISHED) asterisk 2878 asterisk 528u IPv4 29757779 0t0 TCP 172.20.255.41:55627-10.200.14.29:sip (ESTABLISHED) asterisk 2878 asterisk 530u IPv4 19211854 0t0 TCP 172.20.255.40: sip-10.100.157.32:49227 (ESTABLISHED) sip show settings Global Settings: UDP Bindaddress:172.20.255.40:5060 TCP SIP Bindaddress:172.20.255.40:5060 Anyone has idea what could be the reason? Regards, Zohair Raza -- _ -- 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
