Re: [asterisk-users] Allow Header
Hi, asterisk version is 1.4.13 rich... On Tue, Nov 10, 2009 at 7:01 AM, Tilghman Lesher tles...@digium.com wrote: On Monday 09 November 2009 15:38:54 Coco Richard wrote: i'm not sure to understand. Asterisk does support SIP INFO, so why doesn't Asterisk add the INFO Method in the 200OK Response? You must be using Asterisk 1.2. This is the only version that I could find that does not put the INFO tag into the Allow header. Asterisk 1.4 and all versions greater supply the INFO tag as standard. Given that 1.2 is in security-only fix mode now, this is not going to be changed in SVN or in any subsequent 1.2 release (if any). You're welcome to change the ALLOWED_METHODS define in the top of chan_sip.c and recompile, however. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call audio leaking between calls
Hi Has anyone ever had experience of phones on the same office network being able to hear other concurrent call's audio whilst on calls of their own? We're getting this for the first time and I'm at a bit of a loss as to where to start to look. We're using 1.4.17 Any pointers would be much appreciated! Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call audio leaking between calls
Aslamoalikum Ishfaq Can you check this with asterisk 1.6.X ? On Tue, Nov 10, 2009 at 2:45 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Has anyone ever had experience of phones on the same office network being able to hear other concurrent call's audio whilst on calls of their own? We're getting this for the first time and I'm at a bit of a loss as to where to start to look. We're using 1.4.17 Any pointers would be much appreciated! Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allow Header
I took a look in chan_sip.c an for 1.4.13 ALLOWED_METHODS doesn't add INFO. So I will upgrade to 1.6... thank you for the replies... rich... On Tue, Nov 10, 2009 at 9:21 AM, Coco Richard richard.kingc...@gmail.com wrote: Hi, asterisk version is 1.4.13 rich... On Tue, Nov 10, 2009 at 7:01 AM, Tilghman Lesher tles...@digium.com wrote: On Monday 09 November 2009 15:38:54 Coco Richard wrote: i'm not sure to understand. Asterisk does support SIP INFO, so why doesn't Asterisk add the INFO Method in the 200OK Response? You must be using Asterisk 1.2. This is the only version that I could find that does not put the INFO tag into the Allow header. Asterisk 1.4 and all versions greater supply the INFO tag as standard. Given that 1.2 is in security-only fix mode now, this is not going to be changed in SVN or in any subsequent 1.2 release (if any). You're welcome to change the ALLOWED_METHODS define in the top of chan_sip.c and recompile, however. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call audio leaking between calls
Hi Not really, all our asterisk servers are 1.4.17 and the issue isn't happening to all our customers, only one subset and I wouldn't even know where to start to replicate the issue as our own office network runs off the same server and we've never had this sort of issue. Ish ABBAS SHAKEEL wrote: Aslamoalikum Ishfaq Can you check this with asterisk 1.6.X ? On Tue, Nov 10, 2009 at 2:45 PM, Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote: Hi Has anyone ever had experience of phones on the same office network being able to hear other concurrent call's audio whilst on calls of their own? We're getting this for the first time and I'm at a bit of a loss as to where to start to look. We're using 1.4.17 Any pointers would be much appreciated! Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR Import
Hi, how to write the cdr directly to the databse (Mysq)instead of importing Master.csv to table using a php script. Noting that I load asterisk_addons_mysql rev-xx-xx-xx-xx*CLI cdr status rev-xx-xx-xx-xx*CLI Call Detail Record (CDR) settings -- Logging:Enabled Mode: Simple Log unanswered calls: Yes * Registered Backends --- csv cdr_sqlite3_custom cdr-custom regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Import
Khaled W Chehab schrieb: how to write the cdr directly to the databse (Mysq)instead of importing Master.csv to table using a php script. Noting that I load asterisk_addons_mysql cdr_mysql from Asterisk Addons. Configuration file: /etc/asterisk/cdr_mysql.conf Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call audio leaking between calls
Ishfaq Malik wrote: Has anyone ever had experience of phones on the same office network being able to hear other concurrent call's audio whilst on calls of It's called cross talk and yes, we've experienced it. But, it will only happen on an analog network (PSTN). At that point, the provider had to check the analog lines. It eventually was fixed. In a purely SIP environment, you shouldn't see this. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIREN14 call setup and record/playback
On Thu, Nov 5, 2009 at 8:23 AM, Kevin P. Fleming kpflem...@digium.comwrote: We need to see how you are originating the calls; it's up to the originator to specify the formats that will be allowed for that call. In spool files, for example, there is a header that can be included to specify which audio (and video) codecs should be offered on the outgoing channel. Thanks Kevin, I was unaware of the Codecs header for the spool file. However Asterisk still appears to be less than satisfied when asked to initiate a call with 'siren14' as the *only* codec. (Obviously it isn't yet a full codec for Asterisk and is only a supported format. I suspect that is the key to this observation) As a clean test, I did the following on a fresh install of CentOS: svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk cd asterisk ./configure make menuselect make install make samples cp /usr/local/src/asterisk/contrib/init.d/rc.redhat.asterisk /etc/init.d/asterisk asterisk -vvv | grep siren == Registered file format siren7, extension(s) siren7 format_siren7.so = (ITU G.722.1 (Siren7, licensed from Polycom)) == Registered file format siren14, extension(s) siren14 format_siren14.so = (ITU G.722.1 Annex C (Siren14, licensed from Polycom)) (first make sure basic spool call works) vi /etc/asterisk/sip.conf disallow=all allow=ulaw service asterisk restart vi call.txt Channel: SIP/f...@bar.com CallerID: testcall Context: default Extension: demo Codecs: ulaw cp call.txt /var/spool/asterisk/outgoing/ Outgoing INVITE sent to the folks at bar.com (now let's try just siren14) vi /etc/asterisk/sip.conf disallow=all allow=siren14 service asterisk restart vi call.txt Channel: SIP/f...@bar.com CallerID: testcall Context: default Extension: demo Codecs: siren14 cp call.txt /var/spool/asterisk/outgoing/ -- Attempting call on SIP/f...@bar.com for d...@default:1 (Retry 1) [Nov 10 07:43:13] WARNING[27630]: chan_sip.c:5735 sip_call: No audio format found to offer. Cancelling call to foo So while inbound calls work fine with siren14 as the only allow=, Asterisk won't initiate an outbound call with siren14 as the only choice. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call audio leaking between calls
Doug Lytle wrote: Ishfaq Malik wrote: Has anyone ever had experience of phones on the same office network being able to hear other concurrent call's audio whilst on calls of It's called cross talk and yes, we've experienced it. But, it will only happen on an analog network (PSTN). At that point, the provider had to check the analog lines. It eventually was fixed. In a purely SIP environment, you shouldn't see this. Doug This is what I'm thinking too and it's a weird one to try to pin down, especially as I've currently got very little information. I think I'm going to use Monitor on all their calls and see if the recordings show any signs of this cross talk but even if they do it still doesn't help to resolve the issue. You'd think it would be an impossibility due to the nature of IP traffic. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call audio leaking between calls
Ishfaq Malik wrote: You'd think it would be an impossibility due to the nature of IP traffic. I've also read, on this list, that some have had issues with people using speaker phones and the far end picking up conversations in a crowded call center environment. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call audio leaking between calls
I agree with the cross talk analysis. My suggestion would be to focus your efforts on the analog trunks/stations, not SIP. Are you using twisted pair or shielded cables for your analog runs? If not, you might consider changing the cables or at least increasing the physical distance between them - in my experience this is the most common cause for cross talk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Tuesday, November 10, 2009 7:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call audio leaking between calls Doug Lytle wrote: Ishfaq Malik wrote: Has anyone ever had experience of phones on the same office network being able to hear other concurrent call's audio whilst on calls of It's called cross talk and yes, we've experienced it. But, it will only happen on an analog network (PSTN). At that point, the provider had to check the analog lines. It eventually was fixed. In a purely SIP environment, you shouldn't see this. Doug This is what I'm thinking too and it's a weird one to try to pin down, especially as I've currently got very little information. I think I'm going to use Monitor on all their calls and see if the recordings show any signs of this cross talk but even if they do it still doesn't help to resolve the issue. You'd think it would be an impossibility due to the nature of IP traffic. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] looking for an Asterisk supervision (status viewer) tool
Hi! I am looking for a tool (application or webinterface) which shows me the current status of an Asterisk server, e.g.: - Status of the SIP peers (registered/offline) - current incoming and outgoing calls - start-time, numbers, some history - history (calls stopped in the last 15 minutes, who hang up?) - should be possible to link those calls to the relevant SIP peers - kill calls Before coding it myself, is there something you can recommend to me? The thing should be complete auto configured, e.g. no configuration file which peers/channels to be displayed, just fetch all the configuration from Asterisk and display it. thanks klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call audio leaking between calls
The analogue part of it is being supplied by an external carrier. If the problem were with them then the cross talk would be able to happen to any of THEIR customers and not just a sub set of ours. But thanks to all for all the ideas and insight, it helps me realise I'm not going potty with my original thoughts... Ish Ryan M. Colbert wrote: I agree with the cross talk analysis. My suggestion would be to focus your efforts on the analog trunks/stations, not SIP. Are you using twisted pair or shielded cables for your analog runs? If not, you might consider changing the cables or at least increasing the physical distance between them - in my experience this is the most common cause for cross talk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Tuesday, November 10, 2009 7:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call audio leaking between calls Doug Lytle wrote: Ishfaq Malik wrote: Has anyone ever had experience of phones on the same office network being able to hear other concurrent call's audio whilst on calls of It's called cross talk and yes, we've experienced it. But, it will only happen on an analog network (PSTN). At that point, the provider had to check the analog lines. It eventually was fixed. In a purely SIP environment, you shouldn't see this. Doug This is what I'm thinking too and it's a weird one to try to pin down, especially as I've currently got very little information. I think I'm going to use Monitor on all their calls and see if the recordings show any signs of this cross talk but even if they do it still doesn't help to resolve the issue. You'd think it would be an impossibility due to the nature of IP traffic. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup
Hi, is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten = h,n,HangUp(channelname) it doesn't hangup... Is that correct? Thanks, _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup, SoftHangup
Anahi Ludueña schrieb: is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten = h,n,HangUp(channelname) it doesn't hangup... Is that correct? You need to use the SoftHangup() application. core show application SoftHangup Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup, SoftHangup
Thanks Phillipp!, it works! Anahi Ludueña Date: Tue, 10 Nov 2009 14:44:09 +0100 From: philipp.kemp...@amooma.de To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hangup, SoftHangup Anahi Ludueña schrieb: is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten = h,n,HangUp(channelname) it doesn't hangup... Is that correct? You need to use the SoftHangup() application. core show application SoftHangup Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Sólo hay un loro experto en Windows 7 en todo el mundo. Y vive en Sietes ¡Cónocelo! http://www.sietesunpueblodeexpertos.com/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is voicemail to text possible?
You could try CMU Sphinx http://cmusphinx.sourceforge.net/html/cmusphinx.php The best speech recognition available today is offshore human decoding, unfortunately. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Monday, November 09, 2009 7:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Is voicemail to text possible? Hi, I understand that speech recognition technology is not very reliable, but skype has has launched a voicemail to text service, and googling showed that some other companies are also offering similar services. I haven't used any such service yet, but was curious is there any open source software available, which, to some extent, could help converting speech from voicemial wav files to text files and could be used with Asterisk? Or is there any other way to accomplish this? -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting outgoing callerid on when using a PRI
Hi list. I've googled around for this, and so far have come up short.
I'm attempting to set the outbound callerid when making a call using
our PRI. I've talked with ATT (the providor) and they have said the
screening table has been disabled, so anything my phone system sends
for callerid should be passing along. However, when I make a call,
the primary number assigned on our PRI is showing up, not the value
I've set.
Here is how I'm making the call...
exten = _91NXXNXX,1,Set(CALLERID(all)=Corner Homecare 12703653903)
exten = _91NXXNXX,2,Dial(DAHDI/G2/${EXTEN:1}})
I'll be glad to offer up any debugging information that you might
need, if it will help. I'm using Asterisk 1.6.0.14 and DAHDI 2.2.0.
TIA
-jonathan
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Re: [asterisk-users] Call audio leaking between calls
Hi All, getting somewhere with this and thought I'd share it in case anyone else has the same problem. First of all, thanks to Doug who told me it's called cross talk and thus helped me find solutions. This makes interesting reading about the topic http://forums.digium.com/viewtopic.php?t=4048highlight=sid=b0524e2ff689fe6ae3387a5ee6d7d92c Now there is an issue that the SIP device should reject any packets not specifically destined for it but that's a separate issue. Our customer has a switch but a non configurable one which sounds no better than a hub to me, I think I'm going to have to get him to invest in some better hardware. Ish Ryan M. Colbert wrote: I agree with the cross talk analysis. My suggestion would be to focus your efforts on the analog trunks/stations, not SIP. Are you using twisted pair or shielded cables for your analog runs? If not, you might consider changing the cables or at least increasing the physical distance between them - in my experience this is the most common cause for cross talk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Tuesday, November 10, 2009 7:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call audio leaking between calls Doug Lytle wrote: Ishfaq Malik wrote: Has anyone ever had experience of phones on the same office network being able to hear other concurrent call's audio whilst on calls of It's called cross talk and yes, we've experienced it. But, it will only happen on an analog network (PSTN). At that point, the provider had to check the analog lines. It eventually was fixed. In a purely SIP environment, you shouldn't see this. Doug This is what I'm thinking too and it's a weird one to try to pin down, especially as I've currently got very little information. I think I'm going to use Monitor on all their calls and see if the recordings show any signs of this cross talk but even if they do it still doesn't help to resolve the issue. You'd think it would be an impossibility due to the nature of IP traffic. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble registering Cisco 7942
In your sip.conf file, be sure to specify nat=no for the phone, even though the phone is behind a nat device. The cisco phones handle sip packets differently than the way asterisk expects, so you have to do this in order to make asterisk send the way the phone will accept. Thanks, --Warren Selby On Nov 9, 2009, at 8:35 PM, Stephen Reese rsre...@gmail.com wrote: On Sat, Nov 7, 2009 at 11:36 AM, Warren Selby wcse...@selbytech.com wrote: I think your featureLabel definition is wrong. On the login issue, ssh to the ip of the phone and login first with the user/pass you defined in the file (admin/123), then at the second login prompt use log/log. That should get you the log files which will show you your error. Thanks for the insight. After you mentioned that the syntax of the XML file may be wrong I looked around and found a more complete configuration I could find since mine was a copy and paste special. Using the new configuration the phone comes up but is unable register I *think* it may be an issue with NAT. When the phone fires up for the first time it tries to register for a while and the log didn't help much so I took a peak at the asterisk logging. It seems like packets are not getting back to the phone. I've enabled NAT in the configuration similar to how the other phones are configured but no dice. Note that the Asterisk device is not NATed but the phones are behind a NAT device. I get multiple of the following message in the phone: ERR 16:40:16.273722 JVM: %REG send failure: REGISTER On the asterisk server I keep getting NAT retries: Retransmitting #4 (NAT) to 71.226.175.137:1026: OPTIONS sip:1...@ip of NAT device:1027;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP ASTERISK IP:5060;branch=z9hG4bK53121c03;rport From: asterisk sip:aster...@209.251.157.91;tag=as5b0b32f5 To: sip:1...@ip of NAT:1027;user=phone;transport=udp Contact: sip:aster...@209.251.157.91 Call-ID: 090e1e583f29f9f000dd30ff5719f...@209.251.157.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 10 Nov 2009 02:26:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 Below is the full XML config for the phone: device xsi:type=axl:XIPPhone ctiid=9044468655 deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword123/sshPassword devicePool dateTimeSetting dateTemplateM/D/Ya/dateTemplate timeZoneEastern Standard/Daylight Time/timeZone ntps ntp name192.43.244.18/name ntpModedirectedbroadcast/ntpMode /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeNameAsterisk IP/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies backupProxy/backupProxy backupProxyPort/backupProxyPort emergencyProxy/emergencyProxy emergencyProxyPort/emergencyProxyPort outboundProxyAsterisk IP/outboundProxy outboundProxyPort5060/outboundProxyPort registerWithProxytrue/registerWithProxy /sipProxies sipCallFeatures cnfJoinEnabledtrue/cnfJoinEnabled callForwardURIx--serviceuri-cfwdall/callForwardURI callPickupURIx-cisco-serviceuri-pickup/callPickupURI callPickupListURIx-cisco-serviceuri-opickup/ callPickupListURI callPickupGroupURIx-cisco-serviceuri-gpickup/ callPickupGroupURI meetMeServiceURIx-cisco-serviceuri-meetme/meetMeServiceURI abbreviatedDialURIx-cisco-serviceuri-abbrdial/ abbreviatedDialURI rfc2543Holdfalse/rfc2543Hold callHoldRingback2/callHoldRingback localCfwdEnabletrue/localCfwdEnable semiAttendedTransfertrue/semiAttendedTransfer anonymousCallBlock2/anonymousCallBlock callerIdBlocking2/callerIdBlocking dndControl0/dndControl remoteCcEnabletrue/remoteCcEnable /sipCallFeatures sipStack sipInviteRetx6/sipInviteRetx sipRetx10/sipRetx timerInviteExpires180/timerInviteExpires timerRegisterExpires3600/timerRegisterExpires timerRegisterDelta5/timerRegisterDelta timerKeepAliveExpires120/timerKeepAliveExpires timerSubscribeExpires120/timerSubscribeExpires timerSubscribeDelta5/timerSubscribeDelta timerT1500/timerT1 timerT24000/timerT2 maxRedirects70/maxRedirects remotePartyIDfalse/remotePartyID userInfoNone/userInfo /sipStack autoAnswerTimer1/autoAnswerTimer autoAnswerAltBehaviorfalse/autoAnswerAltBehavior autoAnswerOverridetrue/autoAnswerOverride
Re: [asterisk-users] Call audio leaking between calls
On Tuesday 10 November 2009 09:07:30 Ishfaq Malik wrote: Our customer has a switch but a non configurable one which sounds no better than a hub to me, I think I'm going to have to get him to invest in some better hardware. Many consumer-grade switches effectively turn into hubs when more than 1023 MAC addresses are seen on a network. This may be done intentionally by somebody attempting to eavesdrop on all network connections sent through the switch. A reboot of the switch might (temporarily) remedy the problem, but you'd be better off getting an enterprise-grade switch that does not exhibit such misbehavior. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
use pri debugging (pri debug span 1) to verify if the data sent on the
PRI line is correct! (e.g. type on number, ...)
verify with an incoming call and set the same format on outgoing calls.
regards
klaus
Jon Moore schrieb:
Hi list. I've googled around for this, and so far have come up short.
I'm attempting to set the outbound callerid when making a call using
our PRI. I've talked with ATT (the providor) and they have said the
screening table has been disabled, so anything my phone system sends
for callerid should be passing along. However, when I make a call,
the primary number assigned on our PRI is showing up, not the value
I've set.
Here is how I'm making the call...
exten = _91NXXNXX,1,Set(CALLERID(all)=Corner Homecare 12703653903)
exten = _91NXXNXX,2,Dial(DAHDI/G2/${EXTEN:1}})
I'll be glad to offer up any debugging information that you might
need, if it will help. I'm using Asterisk 1.6.0.14 and DAHDI 2.2.0.
TIA
-jonathan
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Re: [asterisk-users] Setting outgoing callerid on when using a PRI
On Tue, Nov 10, 2009 at 10:39 AM, Klaus Darilion klaus.mailingli...@pernau.at wrote: use pri debugging (pri debug span 1) to verify if the data sent on the PRI line is correct! (e.g. type on number, ...) verify with an incoming call and set the same format on outgoing calls. This is what I see on an incomming call (hopefully, this is right part of the pri debug Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '8124307660' ] And on my outgoing call.. Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '2703653903' ] I changed the line in extensions.conf to be SET(CALLERID(ALL)=Corner Homecare 2703653903) To match what it appears I'm getting from ATT, only the 10 digit number. Not sure if it's relevant here, but in /etc/asterisk/chan_dahdi.conf I have the following: pridialplan=unknown prilocaldialplan=national -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters
Hello, 1. How can specify in /etc/dahdi/genconf_parameters file that a port from a B410P board is to be disabled. Playing with comments (see bellow) doesn't help : file /etc/asterisk/dahdi-channels.conf is filled with 4 ports data. pri_termtype SPAN/1 TE SPAN/2 TE SPAN/3 TE # SPAN/4 TE 2. How can specify groups in /etc/dahdi/genconf_parameters ? I would like to group SPAN/1 and SPAN/2 into group 1 and SPAN/3 into group 2. I was unsuccessful with : group_lines 1 pri_termtype SPAN/1 TE SPAN/2 TE group_lines 2 pri_termtype SPAN/3 TE 3. After a dahdi_genconf commande, generated /etc/asterisk/dahdi-channels.conf is like this : ; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS group=5,11 context=remote switchtype = euroisdn signalling = bri_cpe_ptmp channel = 1-2 context = default group = 63 I can see 2 group= and context= lines. What is the difference between them ? Shall I care to have them both ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gradstream Budge Tone-201
bilal ghayyad wrote: Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? We use a couple of BT200 (BT201 with additional ethernet port) for two years now with asterisk, no problems so far, probably you have a bad one. Just what comes to my mind: Cons: 7seg numbers only display! The display is difficult to see at common desk angle Speakerphone could be better Default ringtones are horrible, (you can change them with a little bit of work tough) There isnt any kind of built-in dialplan, you must use an overlapped dialplan (484 response) Pros: Cheap! Audio is OK, no noises or hiss. Looks like a phone, good buttons. Some phones are just too fancy. Headset plug. IMHO, good price/quality relation, dont expect too much from one of the cheapest SIP phones in the market. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
Jon Moore wrote: I changed the line in extensions.conf to be SET(CALLERID(ALL)=Corner Homecare2703653903) To match what it appears I'm getting from ATT, only the 10 digit number. We've got ATT out of the Detroit area, you can't set callerid name, only number. So, try: exten = _91NXXNXX,1,Set(CALLERID(number)=12703653903) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
On Tue, Nov 10, 2009 at 11:43 AM, Doug Lytle supp...@drdos.info wrote: Jon Moore wrote: I changed the line in extensions.conf to be SET(CALLERID(ALL)=Corner Homecare2703653903) To match what it appears I'm getting from ATT, only the 10 digit number. We've got ATT out of the Detroit area, you can't set callerid name, only number. So, try: exten = _91NXXNXX,1,Set(CALLERID(number)=12703653903) I noticed I had the number wrapped in angle brackets here, and removed those as well. Still having the issue though. Thanks for the pointer. Did you have to provide ATT with a list of numbers you would be setting, or does it Just Work? -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is voicemail to text possible?
I agree with you regarding offshore human decoding, and wondering if Skype is doing the same. If you get voicemail converted to text in about 10-20 minute, most probably it is a human listening and typing it. To hire a few people to do it in some cheap part of the world where they can understand and write English for a few cents is entirely possible. Zeeshan On Tue, Nov 10, 2009 at 9:17 AM, Danny Nicholas da...@debsinc.com wrote: You could try CMU Sphinx http://cmusphinx.sourceforge.net/html/cmusphinx.php The best speech recognition available today is offshore human decoding, unfortunately. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Monday, November 09, 2009 7:29 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Is voicemail to text possible? Hi, I understand that speech recognition technology is not very reliable, but skype has has launched a voicemail to text service, and googling showed that some other companies are also offering similar services. I haven't used any such service yet, but was curious is there any open source software available, which, to some extent, could help converting speech from voicemial wav files to text files and could be used with Asterisk? Or is there any other way to accomplish this? -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
If all else fails, there's always spoofing. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Moore Sent: Tuesday, November 10, 2009 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Setting outgoing callerid on when using a PRI On Tue, Nov 10, 2009 at 11:43 AM, Doug Lytle supp...@drdos.info wrote: Jon Moore wrote: I changed the line in extensions.conf to be SET(CALLERID(ALL)=Corner Homecare2703653903) To match what it appears I'm getting from ATT, only the 10 digit number. We've got ATT out of the Detroit area, you can't set callerid name, only number. So, try: exten = _91NXXNXX,1,Set(CALLERID(number)=12703653903) I noticed I had the number wrapped in angle brackets here, and removed those as well. Still having the issue though. Thanks for the pointer. Did you have to provide ATT with a list of numbers you would be setting, or does it Just Work? -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to configure softphones in asterisk
Hi, hi all.Iam new to VOIP so plz forgive me on asking stupid questions. I have installed Asterisk on Centos 5.3, and dowloaded X-LITE softphone on two windows machine. now i want to start from very basic scenario, i want to make two X-Lite phones communicate through asterisk. guys plz plz tell me what r those impt files in asterisk so that both softphone (X-Lite) should start communicating.any support and guidance will be highly appreciated. Regards, Pawan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Libpri-1.4.10.2 Released
Question about the proper way to update LibPRI: 'Bouncing' asterisk after an installing the new LibPRI version does indeed reflect the update: asterisk*CLI pri show version libpri version: 1.4.10.2 BUT some friendly chaps in the IRC channel have suggested that Asterisk Dahdi need to be recompiled as well. Any truth to this? Thanks -Karl On Tue, Oct 20, 2009 at 3:50 PM, Asterisk Development Team asteriskt...@digium.com wrote: The Asterisk Development team is pleased to announce the release of libpri 1.4.10.2, which is available for immediate download at: http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4.10.2.tar.gz This release resolves various issues found in libpri 1.4.10.1 and earlier versions related to scheduler events not being deleted and new ones being created on top of them. This can cause the scheduler to be overfilled, as well as other Q.921 related badness because of runaway scheduled events. Note, this can only happen when Q.931 messages are attempted to be sent during a D-Channel state transient (D-Channel goes down and back up). For a full list of changes in this release, please see the ChangeLog: http://svn.asterisk.org/svn/libpri/tags/1.4.10.2/ChangeLog Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIREN14 call setup and record/playback
Tom Browning wrote: vi call.txt Channel: SIP/f...@bar.com mailto:f...@bar.com CallerID: testcall Context: default Extension: demo Codecs: siren14 cp call.txt /var/spool/asterisk/outgoing/ -- Attempting call on SIP/f...@bar.com mailto:f...@bar.com for d...@default:1 (Retry 1) [Nov 10 07:43:13] WARNING[27630]: chan_sip.c:5735 sip_call: No audio format found to offer. Cancelling call to foo Please run this test with the 'debug' level enabled for the 'console' channel in logger.conf, and then ensure that you have 'core set verbose 10' and 'core set debug 10' before attempting the outbound call. This should give us some information about why chan_sip did not allow the channel to be created. I suspect it may be because your defined peer for bar.com was not actually used, since your spool file has mailto:f...@bar.com in the Channel header, since that is not valid syntax. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to configure softphones in asterisk
aster...@opensourcesolution.in wrote: Hi, hi all.Iam new to VOIP so plz forgive me on asking stupid questions. I have installed Asterisk on Centos 5.3, and dowloaded X-LITE softphone on two windows machine. now i want to start from very basic scenario, i want to make two X-Lite phones communicate through asterisk. guys plz plz tell me what r those impt files in asterisk so that both softphone (X-Lite) should start communicating.any support and guidance will be highly appreciated. Regards, Pawan Would you please stop sending this and just do the research. May I suggest: Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org Everything you need for this is in this book. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to configure softphones in asterisk
Try: http://www.the-asterisk-book.com/unstable/ aster...@opensourcesolution.in wrote: Hi, hi all.Iam new to VOIP so plz forgive me on asking stupid questions. I have installed Asterisk on Centos 5.3, and dowloaded X-LITE softphone on two windows machine. now i want to start from very basic scenario, i want to make two X-Lite phones communicate through asterisk. guys plz plz tell me what r those impt files in asterisk so that both softphone (X-Lite) should start communicating.any support and guidance will be highly appreciated. Regards, Pawan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
Jon Moore wrote: Did you have to provide ATT with a list of numbers you would be setting, or does it Just Work? In our Detroit and Indianpolis installs, it just worked. In our Battle Creek installs, we can only set the numbers to DIDs that we directly own. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
Jon Moore wrote: On Tue, Nov 10, 2009 at 11:43 AM, Doug Lytlesupp...@drdos.info wrote: Jon Moore wrote: Here is what I have in our zaptel.conf span=1,1,0,esf,b8zs defaultzone=us loadzone=us bchan=1-23 dchan=24 And our zapata.conf: switchtype=national context=pri signalling=pri_cpe group=1 echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=-1.0 txgain=-4.0 busydetect=no callprogress=no pridialplan=unknown usercallerid=yes callerid=asreceived channel = 1-23 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
Name is controlled by Telco; you publish new number and Telco will provide name or unknown based on new number. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Tuesday, November 10, 2009 11:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Setting outgoing callerid on when using a PRI Jon Moore wrote: I changed the line in extensions.conf to be SET(CALLERID(ALL)=Corner Homecare2703653903) To match what it appears I'm getting from ATT, only the 10 digit number. We've got ATT out of the Detroit area, you can't set callerid name, only number. So, try: exten = _91NXXNXX,1,Set(CALLERID(number)=12703653903) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for an Asterisk supervision (status viewer) tool
Hi Klaus, Yes all the below is possible/easy with the OrderlyStats call centre management and reporting tool. It's a free download - please see http://www.orderlyq.com/orderlystats.html Kind regards, Christina Casey Accounts Manager Orderly Software Ltd. Subject: [asterisk-users] looking for an Asterisk supervision (status viewer) tool From: Klaus Darilion klaus.mailingli...@pernau.at Date: Tue, 10 Nov 2009 14:04:16 +0100 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi! I am looking for a tool (application or webinterface) which shows me the current status of an Asterisk server, e.g.: - Status of the SIP peers (registered/offline) - current incoming and outgoing calls - start-time, numbers, some history - history (calls stopped in the last 15 minutes, who hang up?) - should be possible to link those calls to the relevant SIP peers - kill calls Before coding it myself, is there something you can recommend to me? The thing should be complete auto configured, e.g. no configuration file which peers/channels to be displayed, just fetch all the configuration from Asterisk and display it. thanks klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
Jon Moore wrote: On Tue, Nov 10, 2009 at 11:43 AM, Doug Lytle supp...@drdos.info wrote: Jon Moore wrote: I changed the line in extensions.conf to be SET(CALLERID(ALL)=Corner Homecare2703653903) To match what it appears I'm getting from ATT, only the 10 digit number. We've got ATT out of the Detroit area, you can't set callerid name, only number. So, try: exten = _91NXXNXX,1,Set(CALLERID(number)=12703653903) I noticed I had the number wrapped in angle brackets here, and removed those as well. Still having the issue though. Thanks for the pointer. Did you have to provide ATT with a list of numbers you would be setting, or does it Just Work? -jonathan I have an ATT PRI out of Holland and it just works. I actually have it pass the internal extension number as outbound called ID (which is 4 digits) when anyone calls my cell phone so I know how to answer and it works fine. This is what I'm using: set(CALLERID(num)=1234567890) Note num and not number I don't know if that was a change from 1.4 to 1.6 or if Doug mistyped it. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Libpri-1.4.10.2 Released
Karl Fife wrote: Question about the proper way to update LibPRI: 'Bouncing' asterisk after an installing the new LibPRI version does indeed reflect the update: asterisk*CLI pri show version libpri version: 1.4.10.2 BUT some friendly chaps in the IRC channel have suggested that Asterisk Dahdi need to be recompiled as well. Any truth to this? No. DAHDI is not related to libpri at all. Asterisk would only need to be recompiled if there were libpri API changes, or to take advantage of new features in libpri, or if you build Asterisk with libpri statically linked in (which would be very uncommon). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Libpri-1.4.10.2 Released
To my knowledge DAHDI does not use libpri, only asterisk. In my experience you can upgrade libpri and restart asterisk, just like you did, to make the upgrade take effect. As to what the proper thing to do is, it's probably better to recompile asterisk after upgrading libpri. -Dave Karl Fife wrote: Question about the proper way to update LibPRI: 'Bouncing' asterisk after an installing the new LibPRI version does indeed reflect the update: asterisk*CLI pri show version libpri version: 1.4.10.2 BUT some friendly chaps in the IRC channel have suggested that Asterisk Dahdi need to be recompiled as well. Any truth to this? Thanks -Karl On Tue, Oct 20, 2009 at 3:50 PM, Asterisk Development Team asteriskt...@digium.com wrote: The Asterisk Development team is pleased to announce the release of libpri 1.4.10.2, which is available for immediate download at: http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4.10.2.tar.gz This release resolves various issues found in libpri 1.4.10.1 and earlier versions related to scheduler events not being deleted and new ones being created on top of them. This can cause the scheduler to be overfilled, as well as other Q.921 related badness because of runaway scheduled events. Note, this can only happen when Q.931 messages are attempted to be sent during a D-Channel state transient (D-Channel goes down and back up). For a full list of changes in this release, please see the ChangeLog: http://svn.asterisk.org/svn/libpri/tags/1.4.10.2/ChangeLog Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
Dave Fullerton wrote: Note num and not number I don't know if that was a change from 1.4 to 1.6 or if Doug mistyped it. Not a mistype. I've been using number all along, but looking at the docs shows that I've been incorrect. It must concatenate the number down to num. Looks like I've got a little modifying to do this evening: core show function CALLERID livonia*CLI -= Info about function 'CALLERID' =- [Syntax] CALLERID(datatype[,optional-CID]) [Synopsis] Gets or sets Caller*ID data on the channel. [Description] Gets or sets Caller*ID data on the channel. The allowable datatypes are all, name, *num*, ANI, DNID, RDNIS. Uses channel callerid by default or optional callerid, if specified. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
Doug Lytle wrote: Dave Fullerton wrote: Note num and not number I don't know if that was a change from 1.4 to 1.6 or if Doug mistyped it. Not a mistype. I've been using number all along, but looking at the docs shows that I've been incorrect. It must concatenate the number down to num. Looks like I've got a little modifying to do this evening: core show function CALLERID livonia*CLI -= Info about function 'CALLERID' =- [Syntax] CALLERID(datatype[,optional-CID]) [Synopsis] Gets or sets Caller*ID data on the channel. [Description] Gets or sets Caller*ID data on the channel. The allowable datatypes are all, name, *num*, ANI, DNID, RDNIS. Uses channel callerid by default or optional callerid, if specified. Doug The documentation is correct, but the way the check really works, is that it reads the first 3 chars and matches it to num. This means that num, number, and numnumnumIloveapplesauce would all technically match. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Libpri-1.4.10.2 Released
No. DAHDI is not related to libpri at all. Asterisk would only need to be recompiled if there were libpri API changes, or to take advantage of new features in libpri, or if you build Asterisk with libpri statically linked in (which would be very uncommon). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ...and presumably any such features API changes would be clearly called out in the libpri ChangeLog? In other words the best current practice would be to check the libpri changelog as an authoritative source for any such changes: If only bug fixes are called out, just bounce asterisk. If you see other changes or features, recompile Asterisk. Is that a fair summary? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
But as Captain Hook would say, using number is bad form . What does not bite us now will eventually get us in a future update. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Tuesday, November 10, 2009 1:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Setting outgoing callerid on when using a PRI Doug Lytle wrote: Dave Fullerton wrote: Note num and not number I don't know if that was a change from 1.4 to 1.6 or if Doug mistyped it. Not a mistype. I've been using number all along, but looking at the docs shows that I've been incorrect. It must concatenate the number down to num. Looks like I've got a little modifying to do this evening: core show function CALLERID livonia*CLI -= Info about function 'CALLERID' =- [Syntax] CALLERID(datatype[,optional-CID]) [Synopsis] Gets or sets Caller*ID data on the channel. [Description] Gets or sets Caller*ID data on the channel. The allowable datatypes are all, name, *num*, ANI, DNID, RDNIS. Uses channel callerid by default or optional callerid, if specified. Doug The documentation is correct, but the way the check really works, is that it reads the first 3 chars and matches it to num. This means that num, number, and numnumnumIloveapplesauce would all technically match. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
Jason Parker wrote: Doug Lytle wrote: Dave Fullerton wrote: Note num and not number I don't know if that was a change from 1.4 to 1.6 or if Doug mistyped it. Not a mistype. I've been using number all along, but looking at the docs shows that I've been incorrect. It must concatenate the number down to num. Looks like I've got a little modifying to do this evening: core show function CALLERID livonia*CLI -= Info about function 'CALLERID' =- [Syntax] CALLERID(datatype[,optional-CID]) [Synopsis] Gets or sets Caller*ID data on the channel. [Description] Gets or sets Caller*ID data on the channel. The allowable datatypes are all, name, *num*, ANI, DNID, RDNIS. Uses channel callerid by default or optional callerid, if specified. Doug The documentation is correct, but the way the check really works, is that it reads the first 3 chars and matches it to num. This means that num, number, and numnumnumIloveapplesauce would all technically match. lol. Love it. I want to use that in my dialplan just to make my successor go WTF? -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
Dave Fullerton wrote: lol. Love it. I want to use that in my dialplan just to make my successor go WTF? I almost lost a mouth full of coffee!! *snicker* Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
I'm gonna put that CALLERID(numWTF) in my dialplan as well... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Tuesday, November 10, 2009 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Setting outgoing callerid on when using a PRI Dave Fullerton wrote: lol. Love it. I want to use that in my dialplan just to make my successor go WTF? I almost lost a mouth full of coffee!! *snicker* Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIREN14 call setup and record/playback
On Tue, Nov 10, 2009 at 1:20 PM, Kevin P. Fleming kpflem...@digium.com wrote: Please run this test with the 'debug' level enabled for the 'console' channel in logger.conf, and then ensure that you have 'core set verbose 10' and 'core set debug 10' before attempting the outbound call. This should give us some information about why chan_sip did not allow the channel to be created. I suspect it may be because your defined peer for bar.com was not actually used, since your spool file has mailto:f...@bar.com in the Channel header, since that is not valid syntax. Sorry, there is no mailto: header in the spool file, that must be gmail parsing my paste as html and adding that format. setting gmail to plain text: Channel: SIP/f...@bar.com CallerID: testcall Context: default Extension: demo Codecs: siren14 The only difference between the call attempt that actually sends the INVITE and the call attempt that complains is 'ulaw' vs 'siren14' in the sip.conf allow= and spol file Codecs: header. Clearly those codec choices are not treated the same to build an outbound INVITE. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIREN14 call setup and record/playback
Tom Browning wrote: The only difference between the call attempt that actually sends the INVITE and the call attempt that complains is 'ulaw' vs 'siren14' in the sip.conf allow= and spol file Codecs: header. Clearly those codec choices are not treated the same to build an outbound INVITE. They are, but we won't be able to know what is happening unless you post a detailed console log like I suggested in my previous reply. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters
On Tue, Nov 10, 2009 at 06:06:12PM +0100, Olivier wrote: Hello, 1. How can specify in /etc/dahdi/genconf_parameters file that a port from a B410P board is to be disabled. There's currently no way to do that. It should be trivial to implment. The more difficult part of it would be how to define exactly what spans / channels to disable. But why do you need that? Playing with comments (see bellow) doesn't help : file /etc/asterisk/dahdi-channels.conf is filled with 4 ports data. pri_termtype SPAN/1 TE SPAN/2 TE SPAN/3 TE # SPAN/4 TE Currently pri_termtype is the only directive in dahdi_genconf that uses this list syntax. I'm not very happy with it. I'm not exactly sure if there should be some sort of generic way of adding per-span (span? channel? how do you define a span?) definitions. Think of ssh_config. 2. How can specify groups in /etc/dahdi/genconf_parameters ? I would like to group SPAN/1 and SPAN/2 into group 1 and SPAN/3 into group 2. I was unsuccessful with : group_lines 1 pri_termtype SPAN/1 TE SPAN/2 TE group_lines 2 pri_termtype SPAN/3 TE 3. After a dahdi_genconf commande, generated /etc/asterisk/dahdi-channels.conf is like this : ; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS group=5,11 context=remote switchtype = euroisdn signalling = bri_cpe_ptmp channel = 1-2 context = default group = 63 I can see 2 group= and context= lines. What is the difference between them ? Shall I care to have them both ? The second ones are not really needed. Unless you want to assume less of the configuration below. TODO: implement a [section] syntax there to care even less about such inclusion. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call declined
Thanks !! it works 2009/11/10 Michael Wyres mwy...@cdm.com.au Try: *[tutorial]** exten = 1234,1,Dial(SIP/gianca,10,t)* *exten = 12345,1,Dial(SIP/giusy,10,t*) You want a “/” between SIP and the name of the phone, not an “,”. The “10” refers to the number of seconds you want the phone to ring. The “t” allows the channel to be transferred after pickup – not strictly needed, but I tend to put it in in most instances as generally you’ll want it. For more information on the Dial application, see http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *giancarlo lombardo *Sent:* Tuesday, 10 November 2009 09:03 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Call declined Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined. Below the sip.conf: *[gianca]** type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy]** type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* extension.conf: *[tutorial]** exten = 1234,1,Dial(SIP,gianca)* *exten = 12345,1,Dial(SIP,giusy*) Below the output of SIP debug of IP caller (192.168.1.116) in asterisk *dhcppc0*CLI** --- SIP read from 192.168.1.116:14862 --- INVITE sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport Max-Forwards: 70 Contact: sip:gia...@192.168.1.116:14862 To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 265* *v=0** o=- 6 2 IN IP4 192.168.1.116 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.116 t=0 0 m=audio 5960 RTP/AVP 107 0 8 101 a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv* *-** --- (12 headers 11 lines) --- Sending to 192.168.1.116 : 14862 (NAT) Using INVITE request as basis request - NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.* *--- Reliably Transmitting (no NAT) to 192.168.1.116:14862 ---** SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;received=192.168.1.116;rport=14862 From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 ;tag=as29d2b71c Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY upported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=42ebb35e Content-Length: 0* *** Scheduling destruction of SIP dialog 'NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.' in 32000 ms (Method: INVITE) Found user 'gianca' dhcppc0*CLI --- SIP read from 192.168.1.116:14862 --- ACK sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 ;tag=as29d2b71c From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 ACK Content-Length: 0* *-** --- (7 headers 0 lines) --- dhcppc0*CLI --- SIP read from 192.168.1.116:14862 --- INVITE sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport Max-Forwards: 70 Contact: sip:gia...@192.168.1.116:14862 To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username=gianca,realm=asterisk,nonce=42ebb35e,uri= sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 ,response=8d00b3e1b28ed2e40681a3a9ee410046,algorithm=MD5 User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 265* *v=0** o=- 6 2 IN IP4 192.168.1.116 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.116 t=0 0 m=audio 5960 RTP/AVP 107 0 8 101 a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv* *-**
[asterisk-users] Silent Dialing
Is there a way to disable ringing while dialing? Example, external users come into our IVR, and if they dial certain IVR options, these are sent off to a remote server for call handling ( Dial(SIP/extens...@remoteserver) for example). It rings once, then the remote system picks up. I would like it to be more transparent to the users. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Import
On 11/11/09 12:56 AM, Philipp Kempgen wrote: Khaled W Chehab schrieb: how to write the cdr directly to the databse (Mysq)instead of importing Master.csv to table using a php script. Noting that I load asterisk_addons_mysql cdr_mysql from Asterisk Addons. Configuration file: /etc/asterisk/cdr_mysql.conf Also, the status check is cdr mysql status -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] user extension in asterisk GUI
Hi all, I just configured some user in sip.conf and extensions.conf; they works fine. Now I'm trying to do the same with Extensions feature of FreePBXAdministration, but I cannot see what I have done manualy. Is gui working with other an source, How can I access such data ? Thanks in advance for any suggestion. -- Giancarlo Lombardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIREN14 call setup and record/playback
On Tue, Nov 10, 2009 at 3:39 PM, Kevin P. Fleming kpflem...@digium.com wrote: They are, but we won't be able to know what is happening unless you post a detailed console log like I suggested in my previous reply. -- Attempting call on SIP/f...@bar.com for d...@default:1 (Retry 1) [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:24104 sip_request_call: Asked to create a SIP channel with formats: 0x40 (slin) [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:7546 sip_alloc: Allocating new SIP dialog for 5034f492225f4eef5db2149b20ad5...@10.1.1.148 - INVITE (No RTP) [Nov 10 17:32:37] DEBUG[28977]: rtp_engine.c:328 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x969b960' [Nov 10 17:32:37] DEBUG[28977]: res_rtp_asterisk.c:423 ast_rtp_new: Allocated port 12038 for RTP instance '0x969b960' [Nov 10 17:32:37] DEBUG[28977]: rtp_engine.c:337 ast_rtp_instance_new: RTP instance '0x969b960' is setup and ready to go [Nov 10 17:32:37] DEBUG[28977]: res_rtp_asterisk.c:2197 ast_rtp_prop_set: Setup RTCP on RTP instance '0x969b960' [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:5251 do_setnat: Setting NAT on RTP to Off [Nov 10 17:32:37] DEBUG[28977]: acl.c:499 ast_ouraddrfor: Found IP address for this socket [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:3851 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 10.1.1.148:5060 [Nov 10 17:32:37] DEBUG[28977]: frame.c:1235 ast_codec_choose: Could not find preferred codec - Going for the best codec [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6977 sip_new: *** Our native formats are 0x40 (slin) [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6978 sip_new: *** Joint capabilities are 0x40 (slin) [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6979 sip_new: *** Our capabilities are 0x4000 (siren14) [Nov 10 17:32:37] DEBUG[28977]: frame.c:1235 ast_codec_choose: Could not find preferred codec - Going for the best codec [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6980 sip_new: *** AST_CODEC_CHOOSE formats are 0x40 (slin) [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6982 sip_new: *** Our preferred formats from the incoming channel are 0x40 (slin) [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:7010 sip_new: This channel will not be able to handle video. [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:5721 sip_call: Outgoing Call for foo [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:5932 update_call_counter: Updating call counter for outgoing call [Nov 10 17:32:37] WARNING[28977]: chan_sip.c:5735 sip_call: No audio format found to offer. Cancelling call to foo Note that there are no peer definitions used. I'm only setting codec preference in sip.conf and the spool file. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIREN14 call setup and record/playback
Tom Browning wrote: On Tue, Nov 10, 2009 at 3:39 PM, Kevin P. Fleming kpflem...@digium.com wrote: They are, but we won't be able to know what is happening unless you post a detailed console log like I suggested in my previous reply. -- Attempting call on SIP/f...@bar.com for d...@default:1 (Retry 1) [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:24104 sip_request_call: Asked to create a SIP channel with formats: 0x40 (slin) If you've specified the codecs properly in the spool file, then this is clearly a bug, because the channel was requested from chan_sip in signed linear (slin) format, not siren14. Since there is no siren14 codec module available, chan_sip can't provide slin to the Asterisk core and siren14 to the SIP peer, so it fails. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble registering Cisco 7942
On Tue, Nov 10, 2009 at 10:13 AM, Warren Selby wcse...@selbytech.com wrote: In your sip.conf file, be sure to specify nat=no for the phone, even though the phone is behind a nat device. The cisco phones handle sip packets differently than the way asterisk expects, so you have to do this in order to make asterisk send the way the phone will accept. Thanks, --Warren Selby Thanks, as a test I changed both a 7960 and 7942 both to nat=no the latter being the one I'm having trouble registering. The 7960 then was unable to register so I changed it back to nat=yes. When I changed the 7942 to nat=no and disabled registerWithProxy I can get a dial tone but can't dial out due to the following: SIP/2.0 407 Proxy Authentication Required I tried re-enabling the proxy but then I get nothing as before. From the sip.conf I would assume that registerWithProxy would be the same as the realm and the proxy statement for the phone be the same as the domain? realm=ns1.domain.net domain=domain.net Thanks again for any help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP response code 603
dear all, what is the meaning of this *Got SIP response 603 Declined back from XXX.XXX.XXX.XXX* is it asterisk related issue , because sometimes my outgoing calls working fine , and in a day for 2 to 3 hours it gives me this my provider says its all fine there any one know meaning of this regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response code 603
DHAVAL INDRODIYA wrote: dear all, what is the meaning of this *Got SIP response 603 Declined back from XXX.XXX.XXX.XXX* is it asterisk related issue , because sometimes my outgoing calls working fine , and in a day for 2 to 3 hours it gives me this my provider says its all fine there any one know meaning of this It is a SIP failure reply that your provider is sending you periodically. Either there is a legitimate reason that they are denying some of your calls, or there is a problem on their end that they are not aware of and/or themselves do not understand. According to RFC 3261, the precise intended meaning of the 603 Declined error message is: --- 21.6.2 603 Decline The callee's machine was successfully contacted but the user explicitly does not wish to or cannot participate. The response MAY indicate a better time to call in the Retry-After header field. This status response is returned only if the client knows that no other end point will answer the request. --- In practice, 603s are often sent instead of 503 Service Unavailable because, strictly, 6xx-class final responses indicate that no further call attempts should be made by the sending user agent to that destination. This is not true of 4xx and 5xx-class errors. Also, why is your name rendered in all-capital letters? Have you considered becoming Dhaval Indrodiya instead of DHaVAL INDRODIYA? -- Alex -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response code 603
thanks Alex, thanks for your reply, is there any changes needed for resolving this issue , in sip.conf or need to change any dial parameter i currently Use IP-to-IP Dialing with option 'm' , also sometimes i got 503 Service Unavailable, is there any eay to resolve it , if anything then please tell me .out of 100 calls i got these errors in 25 calls. regards On Wed, Nov 11, 2009 at 10:55 AM, Alex Balashov abalas...@evaristesys.comwrote: DHAVAL INDRODIYA wrote: dear all, what is the meaning of this *Got SIP response 603 Declined back from XXX.XXX.XXX.XXX* is it asterisk related issue , because sometimes my outgoing calls working fine , and in a day for 2 to 3 hours it gives me this my provider says its all fine there any one know meaning of this It is a SIP failure reply that your provider is sending you periodically. Either there is a legitimate reason that they are denying some of your calls, or there is a problem on their end that they are not aware of and/or themselves do not understand. According to RFC 3261, the precise intended meaning of the 603 Declined error message is: --- 21.6.2 603 Decline The callee's machine was successfully contacted but the user explicitly does not wish to or cannot participate. The response MAY indicate a better time to call in the Retry-After header field. This status response is returned only if the client knows that no other end point will answer the request. --- In practice, 603s are often sent instead of 503 Service Unavailable because, strictly, 6xx-class final responses indicate that no further call attempts should be made by the sending user agent to that destination. This is not true of 4xx and 5xx-class errors. Also, why is your name rendered in all-capital letters? Have you considered becoming Dhaval Indrodiya instead of DHaVAL INDRODIYA? -- Alex -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response code 603
The problem is with your provider, unless there is something wrong with about 1/4th of your calls - i.e. the destination is unroutable by that provider. DHAVAL INDRODIYA wrote: thanks Alex, thanks for your reply, is there any changes needed for resolving this issue , in sip.conf or need to change any dial parameter i currently Use IP-to-IP Dialing with option 'm' , also sometimes i got 503 Service Unavailable, is there any eay to resolve it , if anything then please tell me .out of 100 calls i got these errors in 25 calls. regards On Wed, Nov 11, 2009 at 10:55 AM, Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote: DHAVAL INDRODIYA wrote: dear all, what is the meaning of this *Got SIP response 603 Declined back from XXX.XXX.XXX.XXX* is it asterisk related issue , because sometimes my outgoing calls working fine , and in a day for 2 to 3 hours it gives me this my provider says its all fine there any one know meaning of this It is a SIP failure reply that your provider is sending you periodically. Either there is a legitimate reason that they are denying some of your calls, or there is a problem on their end that they are not aware of and/or themselves do not understand. According to RFC 3261, the precise intended meaning of the 603 Declined error message is: --- 21.6.2 603 Decline The callee's machine was successfully contacted but the user explicitly does not wish to or cannot participate. The response MAY indicate a better time to call in the Retry-After header field. This status response is returned only if the client knows that no other end point will answer the request. --- In practice, 603s are often sent instead of 503 Service Unavailable because, strictly, 6xx-class final responses indicate that no further call attempts should be made by the sending user agent to that destination. This is not true of 4xx and 5xx-class errors. Also, why is your name rendered in all-capital letters? Have you considered becoming Dhaval Indrodiya instead of DHaVAL INDRODIYA? -- Alex -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
