Re: [asterisk-users] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for public_ip:2049
On 10/13/2010 12:09 AM, Paul Belanger wrote: On Tue, Oct 12, 2010 at 8:08 AM, Jonas Kellensjonas.kell...@telenet.be wrote: [Oct 12 14:03:32] DEBUG[9064] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for public_ip:2049 Is something failing, or is this just informative ? No, this is a debug message. Unless you are trying to solve a problem, you do not need to enable debug messages. Well yes I am debugging at the moment, but I just don't understand every debug-message. The 401 is not received by my IP-phone, but I need to be sure that Asterisk is sending the 401. This message in the debug log tells me Asterisk tries to send a 401. How do I know if Asterisk really succeeds in sending the 401 ?? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] user number in conference
Define aa confrence room num and Syntex is like...
Macro(conference-enter,${EXTEN})
On Mon, Oct 11, 2010 at 6:14 PM, Daniel Knoll dan...@danielknoll.de
wrote:
Hey,
i forgot to ask, how can i get the user number from a caller he is in a
conference, i don't find a variable to us this for the current channel.
Only the command meetme list roomnr shows the usernumber, but i
can't use this output.
why not?
asterisk -rx 'meetme list '
Depending on your version, 1.6 has the concise argument, which
transforms the output into convenient exclamation-point-separated
output.
Then you can send it off to awk -F'!' and pick off the first value.
asterisk -rx 'meetme list concise' | awk -F '!' '{print $1}'
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[asterisk-users] Asterisk Hangup Issue in Ringing State with Incoming call
Hi, I have simulated “Chan phone” driver according to my own driver code and I am able to make internal and external [trunk] Asterisk calls. Only issue I am facing is with hangup in ringing state of incoming call. (1) Make a call from external X-lite to FXS and FXS is in ringing state now (2) Disconnect the caller [X-lite] (3) X-lite sending cancel message to asterisk but hangup callback [phone_hangup] is not invoking in driver. (4) Calle [FXS] is ringing till timeout; as I have timeout case incase of ringing exceeded 30sec. If I disconnect the caller after connection got establish then hangup callback for callee is proper and no issues. I found some links in net regarding this, but solution is not clear. Please suggest me, is any additional asterisk settings are required in driver. Thanks for any replies. Regards, Garge. -- _ -- 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] src_mysql problem
On Tuesday, October 12, 2010 05:31:46 pm Tilghman Lesher wrote: On Tuesday 12 October 2010 08:51:15 Oguzhan Kayhan wrote: Hello, I am using 1.6.2.9-1+b1 asterisk.with cdr_mysql. Everything seems workging correctly except cdr logs. It fills up all data when a call established except src and clid Wht can cause this and where should i check?? I'd check the cdr_mysql.conf file. If you'd aliased away the columns (or created blank staticvalues), that could cause this, although I'm not sure why you would. Hi, the problem seems about failover-0.3 script. There was a setting about calledid 6 I lowered it to 2 and it started to write src to log files. -- _ -- 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] realtime users call problem
Hello, I have a default installation of asterisk 1.6.1.9-2 When i create a user in users.conf via asterisk-gui, calls, voicemail etc works. But if i create a user realtime (and my realtime caching is available too) i can see the realtime user with sip show peers. But, my local dial rules does not work. I can call from realtime user to static users(the ones in users.conf) and if they are not available voicemail activates etc. But when i call a realtime user which is already on peer list i got chan_sip.c:20152 handle_request_invite: Call from '' to extension '' rejected because extension not found in context 'DLPN_WorldcallDial'. And this is when i call a static user (works normal) Executing [6...@dlpn_worldcalldial:1] Macro(SIP/-001e, stdexten,6000,SIP/6000) in new stack This is dlpn_worldcalldial [DLPN_WorldcallDial] include = default include = CallingRule_worldcall include = parkedcalls include = conferences include = ringgroups include = voicemenus include = queues include = voicemailgroups include = directory include = pagegroups include = page_an_extension Thanks a lot if you can tell me what to check -- _ -- 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] realtime users call problem
Check sip_buddies table for the correct context entry. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-13 5:31 AM, Oguzhan Kayhan oguzh...@bilkent.edu.tr wrote: Hello, I have a default installation of asterisk 1.6.1.9-2 When i create a user in users.conf via asterisk-gui, calls, voicemail etc works. But if i create a user realtime (and my realtime caching is available too) i can see the realtime user with sip show peers. But, my local dial rules does not work. I can call from realtime user to static users(the ones in users.conf) and if they are not available voicemail activates etc. But when i call a realtime user which is already on peer list i got chan_sip.c:20152 handle_request_invite: Call from '' to extension '' rejected because extension not found in context 'DLPN_WorldcallDial'. And this is when i call a static user (works normal) Executing [6...@dlpn_worldcalldial:1] Macro(SIP/-001e, stdexten,6000,SIP/6000) in new stack This is dlpn_worldcalldial [DLPN_WorldcallDial] include = default include = CallingRule_worldcall include = parkedcalls include = conferences include = ringgroups include = voicemenus include = queues include = voicemailgroups include = directory include = pagegroups include = page_an_extension Thanks a lot if you can tell me what to check -- _ -- 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] DMTF Mode
Hi, Which DTMF mode do people mostly use? I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for feature usage), the tones arent repeated to the end user. So if I call a company that has a menu system, I can't use the menu. Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sound file debug
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Deneen Sent: Tuesday, October 12, 2010 9:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sound file debug On Tue, Oct 12, 2010 at 4:23 PM, Danny Nicholas da...@debsinc.com wrote: dollars.gsm: data dollars.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Can't be 100% certain on #2, but it must have been right because it works now. Go figure. Isn't WAV wav49 and wav plain old pcm (with the wav header)? -M Yep - still trying to get this all worked out. My IVR system uses sox to piece together and playback files. The whole thing works great with plain .gsm, but is encountering glitches when I try to move to wav/WAV format. -- _ -- 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] DMTF Mode
It depends upon whether you are receiving DTMF or sending, and whether you are using a VoIP protocol or using DAHDI/Zaptel. Could you explain a bit what type of setup you have? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-13 9:15 AM, Dan Journo d...@keshercommunications.com wrote: Hi, Which DTMF mode do people mostly use? I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for feature usage), the tones arent repeated to the end user. So if I call a company that has a menu system, I can't use the menu. Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] DMTF Mode
It depends upon whether you are receiving DTMF or sending, and whether you are using a VoIP protocol or using DAHDI/Zaptel. Sorry about the lack of info. It's a simple SIP only setup. A handful of sip phones, an asterisk server, and a sip provider. The DTMF signals from the sip phones are received by Asterisk because they can access features like *1. The DTMF signal from the called party are received by Asterisk because they can also access features like *1. But, the DTMF tones are not passed through from the Sip Phone to the Called Party. The same happens regardless of whether its an incoming or outgoing call. That means, if any of my users try to call a company with a menu system, they can't select any options. How can I tell if Asterisk is sending the tones through to the provider? I need to find out whether its something I'm doing, or something the provider is doing. Any ideas? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
Then that likely means your phone have the correct dtmfmode, but the link between you and the provider doesn't. Make sure both you and the provider are using the same dtmfmode. My experience shows that sometimes it's also between your provider and THEIR provider, and sometimes reporting the issue to them helps. But of course, verify on your side first. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, October 13, 2010 10:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode It depends upon whether you are receiving DTMF or sending, and whether you are using a VoIP protocol or using DAHDI/Zaptel. Sorry about the lack of info. It's a simple SIP only setup. A handful of sip phones, an asterisk server, and a sip provider. The DTMF signals from the sip phones are received by Asterisk because they can access features like *1. The DTMF signal from the called party are received by Asterisk because they can also access features like *1. But, the DTMF tones are not passed through from the Sip Phone to the Called Party. The same happens regardless of whether its an incoming or outgoing call. That means, if any of my users try to call a company with a menu system, they can't select any options. How can I tell if Asterisk is sending the tones through to the provider? I need to find out whether its something I'm doing, or something the provider is doing. Any ideas? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
I just tried this:- [test_calls] exten = 555,1,Answer() exten = 555,n,SendDTMF(12345) exten = 555,n,Playback(beep) I dialed 555 on the sip phone, nothing was heard, and then a beep... It seems that Asterisk isn't sending DTMF. Its only able to receive. Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
Then that likely means your phone have the correct dtmfmode, but the link between you and the provider doesn't. Just carried out another test to see if my provider was working properly:- exten = INCOMINGDDI,1,Wait(1) exten = INCOMINGDDI,n,Answer() exten = INCOMINGDDI,n,SendDTMF(12345) If I dial the incoming number from a normal phone. The DDI comes from my provider. When the calls is answered by asterisk, the tones are played and I can hear them. So it doesnt seem to be a problem with the connection to my provider. -- _ -- 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] DMTF Mode
Can you send us the SIP config of the sip provider (in sip.conf), removing appropriate passwords and static IPs of course. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, October 13, 2010 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode I just tried this:- [test_calls] exten = 555,1,Answer() exten = 555,n,SendDTMF(12345) exten = 555,n,Playback(beep) I dialed 555 on the sip phone, nothing was heard, and then a beep... It seems that Asterisk isn't sending DTMF. Its only able to receive. Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
Can you send us the SIP config of the sip provider (in sip.conf), removing appropriate passwords and static IPs of course. [provider] type=friend host=removed username=removed fromuser=removed secret=password context=incoming_calls dtmfmode=rfc2833 also tried auto. disallow=all allow=gsm allow=ulaw insecure=invite canreinvite=no The provider has confirmed that they support rfc2833 or inband with the right codecs. -- _ -- 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] DMTF Mode
* The provider has confirmed that they support rfc2833 or inband with the right codecs. Are you using the .gsm codec or some other flavor (ulaw, alaw, G729?) -- _ -- 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] DMTF Mode
Are you using the .gsm codec or some other flavor (ulaw, alaw, G729?) This is from the sip.conf for the provider: allow=gsm allow=ulaw This is from the sip extension:- alaw,ulaw,gsm -- _ -- 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] DMTF Mode
I would suggest first to make sure that asterisk is receiving DTMF fine from your IP devices/phones. Do you have a test IVR where you can dial and press digits and verify that asterisk is responding? Once you are sure that asterisk is receiving DTMF fine, then you should ask your provider what DTMF setting you should have on your system. Usually all of them support RFC2833, so if in your sip.conf where you have defined the trunk, dtmfmode is set to rfc2833, your provider should receive it and pass on to the next carrier or trunk. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-13 10:19 AM, Dan Journo d...@keshercommunications.com wrote: It depends upon whether you are receiving DTMF or sending, and whether you are using a VoIP protoc... Sorry about the lack of info. It's a simple SIP only setup. A handful of sip phones, an asterisk server, and a sip provider. The DTMF signals from the sip phones are received by Asterisk because they can access features like *1. The DTMF signal from the called party are received by Asterisk because they can also access features like *1. But, the DTMF tones are not passed through from the Sip Phone to the Called Party. The same happens regardless of whether its an incoming or outgoing call. That means, if any of my users try to call a company with a menu system, they can't select any options. How can I tell if Asterisk is sending the tones through to the provider? I need to find out whether its something I'm doing, or something the provider is doing. Any ideas? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] DMTF Mode
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, October 13, 2010 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode Are you using the .gsm codec or some other flavor (ulaw, alaw, G729?) This is from the sip.conf for the provider: allow=gsm allow=ulaw This is from the sip extension:- alaw,ulaw,gsm Based on this, your call is probably getting to the provider as ulaw (the alaw is thrown out since it isn't in both selections; if you are in U.S. you don't need the alaw). Try the call with higher debug (at least 5) and verify which one is being selected. -- _ -- 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] GXP-21XX
Anyone used the new Grandstream GXP-21XX series phones. We have been testing these phones and like what we see. We are looking for a greater cross section of testing before we roll them to production. Any feed back would be appreciated. We are talking with Grandstream engineering and they are looking for feed back as well. Any input is appreciated. Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
I would suggest first to make sure that asterisk is receiving DTMF fine from your IP devices/phones. Do you have a test IVR where you can dial and press digits and verify that asterisk is responding? Made a quick IVR, and its working for both sides of the asterisk (between the provider and asterisk, and between the sip phones and asterisk). I think its an issue with DTMF Pass-through. Is there a way to disable DMTF passthrough? Maybe asterisk is blocking the signals from being repeated to the other party? -- _ -- 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] innomedia ATA's
We are testing the innomedia ATA's to possibly replace our current line up of ATA's that we are using. Has anyone used their product? What is their track record on stability, voice quality, DTMF talkoff, T.38 Thanks Bryant From: Zeeshan Zakaria zisha...@gmail.com Sent: Wednesday, October 13, 2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DMTF Mode I would suggest first to make sure that asterisk is receiving DTMF fine from your IP devices/phones. Do you have a test IVR where you can dial and press digits and verify that asterisk is responding? Once you are sure that asterisk is receiving DTMF fine, then you should ask your provider what DTMF setting you should have on your system. Usually all of them support RFC2833, so if in your sip.conf where you have defined the trunk, dtmfmode is set to rfc2833, your provider should receive it and pass on to the next carrier or trunk. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-13 10:19 AM, Dan Journo d...@keshercommunications.com wrote: It depends upon whether you are receiving DTMF or sending, and whether you are using a VoIP protoc... Sorry about the lack of info. It's a simple SIP only setup. A handful of sip phones, an asterisk server, and a sip provider. The DTMF signals from the sip phones are received by Asterisk because they can access features like *1. The DTMF signal from the called party are received by Asterisk because they can also access features like *1. But, the DTMF tones are not passed through from the Sip Phone to the Called Party. The same happens regardless of whether its an incoming or outgoing call. That means, if any of my users try to call a company with a menu system, they can't select any options. How can I tell if Asterisk is sending the tones through to the provider? I need to find out whether its something I'm doing, or something the provider is doing. Any ideas? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] DMTF Mode
Based on this, your call is probably getting to the provider as ulaw (the alaw is thrown out since it isn't in both selections; if you are in U.S. you don't need the alaw). Try the call with higher debug (at least 5) and verify which one is being selected. debug 5 doesnt give me any info regarding the codec. By the way, i'm using asterisk 1.4.36 if that makes any difference. Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
On Wed, Oct 13, 2010 at 10:12 AM, Dan Journo d...@keshercommunications.com wrote: How can I tell if Asterisk is sending the tones through to the provider? I need to find out whether its something I'm doing, or something the provider is doing. You need to enable DTMF logging (logger.conf) and debug an incoming / outgoing call. However, if your DTMF works locally, with asterisk and SIP phones, but does not with your provider. Then I would suspect the issue is with your ITSP, make sure your provider is not converting out-of-band tones to inband, or something like that. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
How can I tell if Asterisk is sending the tones through to the provider? You need to enable DTMF logging (logger.conf) and debug an incoming / outgoing call. Can you understand this? I can see the DTMF signals coming in. I pressed 5 on the normal phone line, and then I pressed 8 on the sip phone. The call is outgoing from the sip phone (through the provider) to a normal land line phone. http://pastebin.com/UNs177LW -- _ -- 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] DMTF Mode
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, October 13, 2010 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode Based on this, your call is probably getting to the provider as ulaw (the alaw is thrown out since it isn't in both selections; if you are in U.S. you don't need the alaw). Try the call with higher debug (at least 5) and verify which one is being selected. debug 5 doesnt give me any info regarding the codec. By the way, i'm using asterisk 1.4.36 if that makes any difference. Thanks Dan I'm on 1.4.30 and this is what I get using debug 5 -- Accepting AUTHENTICATED call from 192.168.xx.xx: requested format = ulaw, requested prefs = (ulaw|gsm|alaw), actual format = gsm, host prefs = (slin|gsm|ulaw|alaw), priority = mine -- _ -- 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] DMTF Mode
Could features.conf be preventing asterisk from repeating the DTMF tones? -- _ -- 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] GXP-21XX
Typically Grandstream 21XX and 20XX is all we've deployed in the past and have had great success with them. I occasionally ( and I mean rarely ) get complaints about calls when on speaker phone, but I think thats more user error than anything else, i've been using them for a couple years now and have had nothing but the best with them. The only quirk that i'm still looking into, is that dang Intercom button. Other than that, Grandstreams are really the way to go IMHO. Side note: We've probably got close to 400 deployed --Matt On Wed, Oct 13, 2010 at 10:43 AM, Bryant Zimmerman brya...@zktech.comwrote: Anyone used the new Grandstream GXP-21XX series phones. We have been testing these phones and like what we see. We are looking for a greater cross section of testing before we roll them to production. Any feed back would be appreciated. We are talking with Grandstream engineering and they are looking for feed back as well. Any input is appreciated. Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] DMTF Mode
I'm on 1.4.30 and this is what I get using debug 5 -- Accepting AUTHENTICATED call from 192.168.xx.xx: requested format = ulaw, requested prefs = (ulaw|gsm|alaw), actual format = gsm, host prefs = (slin|gsm|ulaw|alaw), priority = mine Strange. I dont get that with debug of 5. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, October 13, 2010 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode Could features.conf be preventing asterisk from repeating the DTMF tones? Perhaps. What is your featuredigittimeout value? -- _ -- 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] DMTF Mode
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, October 13, 2010 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode I'm on 1.4.30 and this is what I get using debug 5 -- Accepting AUTHENTICATED call from 192.168.xx.xx: requested format = ulaw, requested prefs = (ulaw|gsm|alaw), actual format = gsm, host prefs = (slin|gsm|ulaw|alaw), priority = mine Strange. I dont get that with debug of 5. Try core set verbose 5 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
What is your featuredigittimeout value? Not used. So default 1000ms. I just added alaw to the provider's list, so now it reads (gsm, ulaw, alaw) and its started working in a fashion. The DTMF tones keep getting stuck. I press a number on the sip phone, and the other party hears a tone. But every few tones, it gets stuck and they hear a long tone of about 3 seconds and then it goes off. Any idea whats happening there? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring Setting up Asterisk
I have followed http://www.asterisk.org/AsteriskNOW-1.5-QuickStart I have installed asterisk in virtualBox for now. I am able to login in to console. Now if I want to create a simple PBX in my local network. like I have 5 machine in my network. I am thinking of assigning each a soft phone.and an extension no now whenever any of these dials extension proper user will be connected on line. At the advance level I want to assign a no so that any one from PSTN can also connect those 5 pc using extension. How to proceed? I am very very newbie to asterisk. Please explain the path stepwise Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
I just added alaw to the provider's list, so now it reads (gsm, ulaw, alaw) and its started working in a fashion. The DTMF tones keep getting stuck. I press a number on the sip phone, and the other party hears a tone. But every few tones, it gets stuck and they hear a long tone of about 3 seconds and then it goes off. Here's the debug log for two DTMF tones. The first was fine. The second got stuck. [2010-10-13 16:25:16] DEBUG[3287]: rtp.c:738 process_rfc2833: - RTP 2833 Event: 0005 (len = 4) [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2129 ast_rtp_change_source: Changing ssrc from 775511001 to 1841818300 due to a source change [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2129 ast_rtp_change_source: Changing ssrc from 381691761 to 1746631866 due to a source change [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:940 ast_rtcp_read: Got RTCP report of 72 bytes [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:738 process_rfc2833: - RTP 2833 Event: 0005 (len = 4) [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:635 create_dtmf_frame: Sending dtmf: 53 (5), at 91.110.53.170 [2010-10-13 16:25:16] DEBUG[18139]: channel.c:4565 ast_generic_bridge: Got DTMF begin on channel (SIP/kesher_201-0381) [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2117 ast_rtp_new_source: Setting the marker bit due to a source update [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2117 ast_rtp_new_source: Setting the marker bit due to a source update [2010-10-13 16:25:16] DEBUG[18139]: channel.c:4882 ast_channel_bridge: Bridge stops bridging channels SIP/kesher_201-0381 and SIP/magrathea-0382 [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2129 ast_rtp_change_source: Changing ssrc from 1841818300 to 455288846 due to a source change [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2129 ast_rtp_change_source: Changing ssrc from 1746631866 to 340402601 due to a source change [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:738 process_rfc2833: - RTP 2833 Event: 0005 (len = 4) [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:738 process_rfc2833: - RTP 2833 Event: 0005 (len = 4) [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:738 process_rfc2833: - RTP 2833 Event: 0005 (len = 4) [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:635 create_dtmf_frame: Sending dtmf: 53 (5), at 91.110.53.170 [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:738 process_rfc2833: - RTP 2833 Event: 0005 (len = 4) [2010-10-13 16:25:16] DEBUG[18139]: channel.c:4565 ast_generic_bridge: Got DTMF end on channel (SIP/kesher_201-0381) [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2117 ast_rtp_new_source: Setting the marker bit due to a source update [2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2117 ast_rtp_new_source: Setting the marker bit due to a source update -- _ -- 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] DMTF Mode
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, October 13, 2010 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode What is your featuredigittimeout value? Not used. So default 1000ms. I just added alaw to the provider's list, so now it reads (gsm, ulaw, alaw) and its started working in a fashion. The DTMF tones keep getting stuck. I press a number on the sip phone, and the other party hears a tone. But every few tones, it gets stuck and they hear a long tone of about 3 seconds and then it goes off. Any idea whats happening there? Thanks Dan Since alaw is sort of making it work, either your SIP provider or your other party is not in U.S. Check this link http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode might want to change rfc2833 to auto. -- _ -- 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] DMTF Mode
Dan Journo d...@keshercommunications.com wrote: Based on this, your call is probably getting to the provider as ulaw (the alaw is thrown out since it isn't in both selections; if you are in U.S. you don't need the alaw). Try the call with higher debug (at least 5) and verify which one is being selected. debug 5 doesnt give me any info regarding the codec. By the way, i'm using asterisk 1.4.36 if that makes any difference. I would suggest log dtmf in your logger.conf and put rtp debug on and see if its sending dtmf. Also call the provider and see if they hear the tones. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
Since alaw is sort of making it work, either your SIP provider or your other party is not in U.S. They arent in the US. Everything is in the UK. -- _ -- 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] DMTF Mode
Since alaw is sort of making it work, either your SIP provider or your other party is not in U.S. I dont understand why the codec should make a difference if im using rfc2833. Could you clear that up for me? -- _ -- 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] DMTF Mode
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, October 13, 2010 10:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode Since alaw is sort of making it work, either your SIP provider or your other party is not in U.S. I dont understand why the codec should make a difference if im using rfc2833. Could you clear that up for me? From what I read, the codec could be trying to switch from rfc2833 to inband during the call, causing the stuck effect. -- _ -- 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] DMTF Mode
From what I read, the codec could be trying to switch from rfc2833 to inband during the call, causing the stuck effect. Any way to prevent that? -- _ -- 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] DMTF Mode
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, October 13, 2010 10:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode From what I read, the codec could be trying to switch from rfc2833 to inband during the call, causing the stuck effect. Any way to prevent that? According to the WIKI, changing rfc2833 to auto in sip.conf should do the trick. -- _ -- 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] GXP-21XX
On Wed, 13 Oct 2010, Bryant Zimmerman wrote: Anyone used the new Grandstream GXP-21XX series phones. We have been testing these phones and like what we see. We are looking for a greater cross section of testing before we roll them to production. Any feed back would be appreciated. We are talking with Grandstream engineering and they are looking for feed back as well. Any input is appreciated. That's the replacement for the GXP2000 - which I've deployed a great many of. Only deployed a small number of GXP2110s as reception console phones though and I've not had issues. Grandstream seem to suffer from buggy early software though, so do check their releases and when you find a stable version - stick to it - although I have to say, all the GXP2000 releases over the past couple of years have been stable, so maybe they're learning :) Gordon -- _ -- 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] DMTF Mode
According to the WIKI, changing rfc2833 to auto in sip.conf should do the trick. Didnt help. I'm contacting the provider to see if they have any ideas. -- _ -- 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] DMTF Mode
Thanks to everyone who helped me on this. Hopefully the provider can sort out the sticking tones now. -- _ -- 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] GXP-21XX
Gordon Thanks for the reply. Grandstream has three new phones that will replace the GXP-20XX series as some point. GXP-2000 - GXP-2100, GXP-2010 - GXP-2110, GXP-2020 - GXP-2120. The GXP-2110 has been released the others appear to be on the cusp of release. We have been testing the GXP-2110 for several months now and are looking to see if there is any one else that has used them in production since their release last month. Are there any other early adopters out there? Based on your reply you have used several of the new GXP-2110's with operators. Have you had any issues with screen display issues. What version of the firmware are you on with them. Thanks Bryant From: Gordon Henderson gordon+aster...@drogon.net Sent: Wednesday, October 13, 2010 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] GXP-21XX On Wed, 13 Oct 2010, Bryant Zimmerman wrote: Anyone used the new Grandstream GXP-21XX series phones. We have been testing these phones and like what we see. We are looking for a greater cross section of testing before we roll them to production. Any feed back would be appreciated. We are talking with Grandstream engineering and they are looking for feed back as well. Any input is appreciated. That's the replacement for the GXP2000 - which I've deployed a great many of. Only deployed a small number of GXP2110s as reception console phones though and I've not had issues. Grandstream seem to suffer from buggy early software though, so do check their releases and when you find a stable version - stick to it - although I have to say, all the GXP2000 releases over the past couple of years have been stable, so maybe they're learning :) Gordon -- _ -- 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] Configuring Setting up Asterisk
On Wed, Oct 13, 2010 at 11:26 AM, Jigar Joshi jiga...@gmail.com wrote: How to proceed? I am very very newbie to asterisk. pabelanger ~book infobot [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Setting up Asterisk
Thanks Paul , I want some quick reference tutorials. On Wed, Oct 13, 2010 at 9:58 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Oct 13, 2010 at 11:26 AM, Jigar Joshi jiga...@gmail.com wrote: How to proceed? I am very very newbie to asterisk. pabelanger ~book infobot [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] checking CDR
Hello Asterisk Community, Is there a way to check in asterisk cdrs and extension forwarded? I mean, i'm calling to a ISDN number, wich goes to extension 8222, but this extension is forwarded to another one, the problem is that in CDRs i am able to see the the first step of the call, but never see the forwarded extension, how can i do that? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Create channel bank with TDMoE
Hi Check this out http://spidermux.com/ On Mon, Oct 11, 2010 at 8:18 PM, Karim Davoodi karimdavo...@gmail.comwrote: Hello, I want to create channel bank in this case: channel bank |-| | FXS,FXO-TDMoE--|--Asterisk |-| How can it? -- _ -- 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 Regards, Godson Gera IVR FreeSWITCH Radius India http://godson.in/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP and ANI
ANI and CID are same in SIP some people use P-Asserted-Identity header to send ANI , but that is not a standard specification just a workaround. -- Thanks Regards, Godson Gera IVR FreeSWITCH Radius India http://godson.in/ On Tue, Oct 12, 2010 at 5:07 AM, JR Richardson jmr.richard...@gmail.comwrote: Hi All, My research indicates ANI is not really supported with SIP Channels or passed between SIP servers, even with setting function CALLERID(ANI). So the only place this applies is on PRI interfaces, when sending calls out a ZAP PRI you can set the ANI to whatever and CID Number to a different whatever so on the other end of the PRI you will receive the two different values? Is this correct or is there a way to set ANI on an outgoing SIP channel (like to a PRI gateway) and the gateway will see a CID Number and a separate ANI and insert that into the ISDN messaging down the PRI? Thanks for any clarification. JR -- JR Richardson Engineering for the Masses -- _ -- 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 disconnects after 20 seconds behind NAT
Hi, I have an asterisk server sitting behind a pfsense firewall, I have successfully configured pfsense for NAT traversal, and clients from the internet can call clients inside the network of asterisk, as well as other clients registered with this asterisk server on the internet. The problem now is when a client from the internet do a call, the call disconnects in 10~20 seconds, but during this period the call goes fine and voice is heard on both ends; But when a client on the same network of asterisk calls another client registered from the internet, the call is established without any issues, and it doesn't disconnect. I have also noticed that when internet clients do calls, and the call is established on both ends, if one of the two parties hang up, the other end isn't notified and the call stays opened at this end. I could provide config files if needed. Please advice about resolving this issue. Ahmed Ossama -- _ -- 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] Unable to specify channel 5: No such device or address
Hi, I am trying to set up two bords on my server: TDM410p(This on is ok) and TE110p. This is my system.conf # Span 1: WCTDM/0 Wildcard TDM410P Board 1 (MASTER) fxsks=1,2,3,4 # Span 2: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 span=1,1,0,cas,hdb3cas=1-15:1101cas=17-31:1101dchan=16loadzone=brdefaultzone=br And this is my chan_dahdi.conf: [channels] #include dahdi-channels.conf ;General optionsusecallerid = yeshidecallerid = nocallwaiting = yesthreewaycalling = yestransfer = yesechocancel = yesechocancelwhenbridged = yesrxgain = 3.0txgain = 3.0 ;FXO Modules group = 1;echocancel = yessignalling = fxs_kscontext = Troncos-Analogicoschannel = 1,2,3,4 ;E1 Modules signalling=mfcr2mfcr2_variant=brmfcr2_get_ani_first=nomfcr2_max_ani=20mfcr2_max_dnis=20mfcr2_category=national_subscribermfcr2_logdir=span1mfcr2_logging=all And my dahdi-chennel.conf ; Span 1: WCTDM/0 Wildcard TDM410P Board 1 (MASTER);;; line=1 WCTDM/0/0 FXSKS (In use)signalling=fxs_kscallerid=asreceivedgroup=0context=from-pstnchannel = 1callerid=group=context=default ;;; line=2 WCTDM/0/1 FXSKS (In use)signalling=fxs_kscallerid=asreceivedgroup=0context=from-pstnchannel = 2callerid=group=context=default ;;; line=3 WCTDM/0/2 FXSKS (In use)signalling=fxs_kscallerid=asreceivedgroup=0context=from-pstnchannel = 3callerid=group=context=default ;;; line=4 WCTDM/0/3 FXSKS (In use)signalling=fxs_kscallerid=asreceivedgroup=0context=from-pstnchannel = 4callerid=group=context=default ; Span 2: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 signalling = mfcr2channel = 1-15,17-31group=1context=Troncos-Digitais In dahdi show status, is shown only the TDM410p board.In dahdi show channels is shown only four analogic trunk If I type: dahdi restart, I see the following messages: [Oct 13 14:36:50] WARNING[930]: chan_dahdi.c:2124 dahdi_open: Unable to specify channel 5: No such device or address This is repeated a lot help!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] checking CDR
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias Sent: Wednesday, October 13, 2010 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] checking CDR Hello Asterisk Community, Is there a way to check in asterisk cdrs and extension forwarded? I mean, i'm calling to a ISDN number, wich goes to extension 8222, but this extension is forwarded to another one, the problem is that in CDRs i am able to see the the first step of the call, but never see the forwarded extension, how can i do that? Thanks! The CDR is only going to record all legs on incoming calls. As you state above, your outgoing call is going to show as one leg regardless of how many bounces it takes. -- _ -- 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] checking CDR
Hi, (Following is for asterisk 1.4) For the forwarded calls, you should see two entries in the cdr, and this is because a forwarded call is actually two separate calls. You have to look in the channel and dstchannel fields of the cdr to match the call ids of the calls to figure out which calls were forwarded. Incoming call's channel value and outgoing call's dstchannel value will be the same, except a comma and digit at the end, showing if it was the first call on that id, second, third or more. I have programmed two billing systems, and this is how I catch forwarded calls and bill them, works perfectly fine. Though it is confusing. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-13 1:21 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello Asterisk Community, Is there a way to check in asterisk cdrs and extension forwarded? I mean, i'm calling to a ISDN number, wich goes to extension 8222, but this extension is forwarded to another one, the problem is that in CDRs i am able to see the the first step of the call, but never see the forwarded extension, how can i do that? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] checking CDR
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias Sent: Wednesday, October 13, 2010 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] checking CDR Hello Asterisk Community, Is there a way to check in asterisk cdrs and extension forwarded? I mean, i'm calling to a ISDN number, wich goes to extension 8222, but this extension is forwarded to another one, the problem is that in CDRs i am able to see the the first step of the call, but never see the forwarded extension, how can i do that? Thanks! The CDR is only going to record all legs on incoming calls. As you state above, your outgoing call is going to show as one leg regardless of how many bounces it takes. The way I have addressed this issue is using flag variables that determine how the call has originated. Inbound calls set one state and outbounds calling checks for that state if it exists we assume that it is either a call forward or a transfer. We then check headers and variables to see what state it is. We then forward the outbound call through a call to LOCAL/customeroutbund/number~trackingvars. This will cause the system to create a sperate channel leg for that part of the call. We have found it to very tricky to get this right for both blind and attended transfers as well as call forwards but you can get very close. We were loosing on 100% of our transfers and forwards and now we are down to 3%-5% of the cases where our method does not work. Or 100% method is to use an additional asterisk box that routes all toll bearing inbound and outbound calls we disable forwards and transfers there. That is where we bill from so we don't loose and $$$. Asterisk 1.8 is looking good with the CEL logging but you have to sift the records to create billing CDR's Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] checking CDR
The real question is are you having the phone forward the calls or is your dial plan redirecting to outbound calling? Bryant From: Zeeshan Zakaria zisha...@gmail.com Sent: Wednesday, October 13, 2010 2:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] checking CDR Hi, (Following is for asterisk 1.4) For the forwarded calls, you should see two entries in the cdr, and this is because a forwarded call is actually two separate calls. You have to look in the channel and dstchannel fields of the cdr to match the call ids of the calls to figure out which calls were forwarded. Incoming call's channel value and outgoing call's dstchannel value will be the same, except a comma and digit at the end, showing if it was the first call on that id, second, third or more. I have programmed two billing systems, and this is how I catch forwarded calls and bill them, works perfectly fine. Though it is confusing. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-13 1:21 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello Asterisk Community, Is there a way to check in asterisk cdrs and extension forwarded? I mean, i'm calling to a ISDN number, wich goes to extension 8222, but this extension is forwarded to another one, the problem is that in CDRs i am able to see the the first step of the call, but never see the forwarded extension, how can i do that? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP disconnects after 20 seconds behind NAT
Am 13.10.2010 19:50, schrieb Ahmed Ossama: Hi, I have an asterisk server sitting behind a pfsense firewall, I have successfully configured pfsense for NAT traversal, and clients from the internet can call clients inside the network of asterisk, as well as other clients registered with this asterisk server on the internet. The problem now is when a client from the internet do a call, the call disconnects in 10~20 seconds, but during this period the call goes fine and voice is heard on both ends; But when a client on the same network of asterisk calls another client registered from the internet, the call is established without any issues, and it doesn't disconnect. I have also noticed that when internet clients do calls, and the call is established on both ends, if one of the two parties hang up, the other end isn't notified and the call stays opened at this end. I could provide config files if needed. Please advice about resolving this issue. Ahmed Ossama Hello ahmed, sounds like the typical SIP ALG problem. Just configure your firewall to do stupid plain nat and dont touch the sip headers. As you could see this doesnt work. if you turn on sip debug you will see several retransmits for the 200 ok message which comes at the real beginning of a call (when you answer the phone) cause the ACK package to this 200 ok could not be received. same to Bye at the end of a call. Best regards Stefan Schmidt -- _ -- 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] call forwarding callerID
Hi list, This is not necessarily an asterisk issue, but a lot of you guys know way more then me, so I have a question: someone at my company sets his phone to forward calls to his cellphone, so someone calls our office, call is forwarded to his cell, and the callerID that shows up on his cell is of course our office number, because asterisk originates a new call to his cell and then bridges the two. so he told me, a partner of his, at his office does the same thing, and when he does it, the callerID shows up as coming from the initial caller, not from his office. so here's the schematic: customer - our office ---callforward-- cellphone so should I call ATT and ask them to unlock our callerID so I can set the outgoing callerID to the customer's number in my dialplan? or is there some other way to handle this? I appreciate any input, Thanks! -- Gerard Saraber Network Admin. Rarcoa, Inc (630) 654-2580 x199 (630) 654-3556 (fax) (630) 915-4122 (cell) -- _ -- 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] GXP-21XX
On Wed, 13 Oct 2010, Bryant Zimmerman wrote: Gordon Thanks for the reply. Grandstream has three new phones that will replace the GXP-20XX series as some point. GXP-2000 - GXP-2100, GXP-2010 - GXP-2110, GXP-2020 - GXP-2120. The GXP-2110 has been released the others appear to be on the cusp of release. We have been testing the GXP-2110 for several months now and are looking to see if there is any one else that has used them in production since their release last month. Are there any other early adopters out there? Based on your reply you have used several of the new GXP-2110's with operators. Have you had any issues with screen display issues. What version of the firmware are you on with them. I've just realised that they're not 2110, but 2020's - they look almost identical (but without the *HD* sticker) which is probably what threw me... But no issues with a few of those - in use in busy offices as an 18-BLF/key reception phone... Been using Grandstreams for over 4 years now - still to find a phone with their features for the price. They're not the best, but have proven to be OK over the years. Gordon Thanks Bryant From: Gordon Henderson gordon+aster...@drogon.net Sent: Wednesday, October 13, 2010 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] GXP-21XX On Wed, 13 Oct 2010, Bryant Zimmerman wrote: Anyone used the new Grandstream GXP-21XX series phones. We have been testing these phones and like what we see. We are looking for a greater cross section of testing before we roll them to production. Any feed back would be appreciated. We are talking with Grandstream engineering and they are looking for feed back as well. Any input is appreciated. That's the replacement for the GXP2000 - which I've deployed a great many of. Only deployed a small number of GXP2110s as reception console phones though and I've not had issues. Grandstream seem to suffer from buggy early software though, so do check their releases and when you find a stable version - stick to it - although I have to say, all the GXP2000 releases over the past couple of years have been stable, so maybe they're learning :) Gordon -- _ -- 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] call forwarding callerID
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard Sent: Wednesday, October 13, 2010 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call forwarding callerID Hi list, This is not necessarily an asterisk issue, but a lot of you guys know way more then me, so I have a question: someone at my company sets his phone to forward calls to his cellphone, so someone calls our office, call is forwarded to his cell, and the callerID that shows up on his cell is of course our office number, because asterisk originates a new call to his cell and then bridges the two. so he told me, a partner of his, at his office does the same thing, and when he does it, the callerID shows up as coming from the initial caller, not from his office. so here's the schematic: customer - our office ---callforward-- cellphone so should I call ATT and ask them to unlock our callerID so I can set the outgoing callerID to the customer's number in my dialplan? or is there some other way to handle this? I appreciate any input, Thanks! -- Gerard Saraber Network Admin. Rarcoa, Inc (630) 654-2580 x199 (630) 654-3556 (fax) (630) 915-4122 (cell) I think FOLLOWME is going to fix this for you -- _ -- 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] call forwarding callerID
On Wed, 13 Oct 2010, Gerard wrote: This is not necessarily an asterisk issue, but a lot of you guys know way more then me, so I have a question: someone at my company sets his phone to forward calls to his cellphone, so someone calls our office, call is forwarded to his cell, and the callerID that shows up on his cell is of course our office number, because asterisk originates a new call to his cell and then bridges the two. so he told me, a partner of his, at his office does the same thing, and when he does it, the callerID shows up as coming from the initial caller, not from his office. so here's the schematic: customer - our office ---callforward-- cellphone so should I call ATT and ask them to unlock our callerID so I can set the outgoing callerID to the customer's number in my dialplan? or is there some other way to handle this? It depends on the technology and the carrier. A simple POTS line and you're out of luck. If you have a T1 (i.e. ISDN-PRI) and a co-operative carrier, it may just work or they may enable it if requested. You could always use a co-operative SIP carrier (like Vitelity). A penny or 2 per minute will keep your someone happy. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] DMTF Mode
Since alaw is sort of making it work, either your SIP provider or your other party is not in U.S. Its stopped working again. This is really unusual. I didnt change anything. I decided to do a tcpdump, and I can clearly see the rfc2833 packets being exchanged correctly. Why should both parties not be able to hear the tones? -- _ -- 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] Some give 603 Declined
Hi, I have some problem with my provider. While the sip registration is successful, i intermittently encounter problem in dialing out. I receive 603 Declined error in my Sjphone client. The asterisk log shows line is busy/congestion. Appreciate if help or direction can be provided. Thanks. CK -- _ -- 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] Some give 603 Declined
On Wed, Oct 13, 2010 at 6:48 PM, asterisk asterisk aster...@ck-lee.com wrote: Appreciate if help or direction can be provided. 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. http://www.ietf.org/rfc/rfc3261.txt Collect a SIP trace and see if a reason is supplied. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] advice re: Page() application
Hi all,I'm planning a new Asterisk installation; the users want to duplicate the paging function they have with their current Panasonic hybrid system. They dial *3 and announce a held call on line 3, for example, and the announcements comes out of all the desktop phone speakers. I'm planning to implement this using the Page() application in addition to parking the call. The O'Reilly book doesn't talk much about Page(), just says that it dumps the channels into a dynamically created MeetMe room which is quickly torn down.To make this work with typical desktop speakerphones, is there anything I need to do in sip.conf? (I was thinking I might need to set autoanswer=yes, for example). I can use a second line presence on all the phones to support this if necessary; I'm using SPA-942s. I don't want all the phones to ring - just have the announcement audible at each phone without the user needing to pick up.I apologize for not being able to try this out myself - I'm out of the country with no access to sip phones right now. Any help/lessons learned using Page() would be most appreciated!Regards,Cassius Smith -- _ -- 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] advice re: Page() application
Hi Cassius, Can`t help for SPA-942, but the Wiki had good info on the Polycoms. Use the Wiki and you`ll do good. Two warnings: 1) It seems to me that the adhoc MeetMe room used by the page application slows things down quite a lot. If you page and have a phone nearby, you`ll hear yourself with quite a bit of delay. It`s very annoying if you`re paging and hearing the page at the same time. Apparently 1.8 supports multicast and will do this differently, but it’ll be a long while before I trust 1.8 to be stable enough for my needs. 2) If you`re doing this over an Internet link (i.e. hosted PBX), keep in mind that because of the MeetMe (I imagine), even if the receiving phones aren’t creating audio, the bandwidth is still is used as if everyone was talking at the same time in a MeetMe room. No biggie if everything is on the LAN, but a bit of a problem if not and you have many phones. And here is a tip: auto-answer is good, but you`ll have to loop through every SIP registration on the phone before using Page() to see if they are being used before adding them to the Page. If not, the phone will not auto-answer (since you`re on a call already) but you`ll have a missed call everytime somebody pages you while you`re on the phone. Users hate that (with reason). You check if each and every phone is being used BEFORE adding them to your page. In other words, if 10 out of 15 phones are idle, Page() only those 10. Besides that, things work as advertised. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Wednesday, October 13, 2010 7:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] advice re: Page() application Hi all, I'm planning a new Asterisk installation; the users want to duplicate the paging function they have with their current Panasonic hybrid system. They dial *3 and announce a held call on line 3, for example, and the announcements comes out of all the desktop phone speakers. I'm planning to implement this using the Page() application in addition to parking the call. The O'Reilly book doesn't talk much about Page(), just says that it dumps the channels into a dynamically created MeetMe room which is quickly torn down. To make this work with typical desktop speakerphones, is there anything I need to do in sip.conf? (I was thinking I might need to set autoanswer=yes, for example). I can use a second line presence on all the phones to support this if necessary; I'm using SPA-942s. I don't want all the phones to ring - just have the announcement audible at each phone without the user needing to pick up. I apologize for not being able to try this out myself - I'm out of the country with no access to sip phones right now. Any help/lessons learned using Page() would be most appreciated! Regards, Cassius Smith -- _ -- 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] MySQL and Channel Event Logging
Hey all, sorry if this has been covered, but I've not found anything after a couple hours' worth of googling. I can see (and I'm familiar with) all the usual MySQL addon apps once I install Asterisk 1.8.x, but I cannot find any reference to MySQL and the new CEL logging tool other than ODBC. Is this the only method available to use MySQL with CEL at this time? Thanks, Sherwood -- _ -- 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] advice re: Page() application
Thanks Mike - this does help. The setup will be a local server on the LAN, and hopefully have plenty of snort to handle the load (20-30 phones). I also am not quite ready to put out 1.8 for my users yet.Do you have a snippet of dialplan code you'd be willing to share to loop through a group and grab/build up a list of channels as you describe? That would be enlightening (and probably save me some time)!What I am hearing is - using a second line presence for the Page() function will work; auto-answer should work and I should only page the phones that are not in use.Cassius Original Message Subject: Re: [asterisk-users] advice re: Page() application From: "Mike" l...@net-wall.com Date: Thu, October 14, 2010 10:12 am To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.com Hi Cassius,Can`t help for SPA-942, but the Wiki had good info on the Polycoms. Use the Wiki and you`ll do good. Two warnings:1) It seems to me that the adhoc MeetMe room used by the page application slows things down quite a lot. If you page and have a phone nearby, you`ll hear yourself with quite a bit of delay. It`s very annoying if you`re paging and hearing the page at the same time. Apparently 1.8 supports multicast and will do this differently, but it’ll be a long while before I trust 1.8 to be stable enough for my needs.2) If you`re doing this over an Internet link (i.e. hosted PBX), keep in mind that because of the MeetMe (I imagine), even if the receiving phones aren’t creating audio, the bandwidth is still is used as if everyone was talking at the same time in a MeetMe room. No biggie if everything is on the LAN, but a bit of a problem if not and you have many phones.And here is a tip: auto-answer is good, but you`ll have to loop through every SIP registration on the phone before using Page() to see if they are being used before adding them to the Page. If not, the phone will not auto-answer (since you`re on a call already) but you`ll have a missed call everytime somebody pages you while you`re on the phone. Users hate that (with reason). You check if each and every phone is being used BEFORE adding them to your page. In other words, if 10 out of 15 phones are idle, Page() only those 10.Besides that, things work as advertised. MikeFrom: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius SmithSent: Wednesday, October 13, 2010 7:54 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] advice re: Page() applicationHi all,I'm planning a new Asterisk installation; the users want to duplicate the paging function they have with their current Panasonic hybrid system. They dial *3 and announce a held call on line 3, for example, and the announcements comes out of all the desktop phone speakers. I'm planning to implement this using the Page() application in addition to parking the call. The O'Reilly book doesn't talk much about Page(), just says that it dumps the channels into a dynamically created MeetMe room which is quickly torn down.To make this work with typical desktop speakerphones, is there anything I need to do in sip.conf? (I was thinking I might need to set autoanswer=yes, for example). I can use a second line presence on all the phones to support this if necessary; I'm using SPA-942s. I don't want all the phones to ring - just have the announcement audible at each phone without the user needing to pick up.I apologize for not being able to try this out myself - I'm out of the country with no access to sip phones right now. Any help/lessons learned using Page() would be most appreciated!Regards,Cassius Smith-- _ -- 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] MySQL and Channel Event Logging
On Wed, Oct 13, 2010 at 9:31 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Hey all, sorry if this has been covered, but I've not found anything after a couple hours' worth of googling. I can see (and I'm familiar with) all the usual MySQL addon apps once I install Asterisk 1.8.x, but I cannot find any reference to MySQL and the new CEL logging tool other than ODBC. Is this the only method available to use MySQL with CEL at this time? Looking at the CEL config files, I don't see one specifically for MySQL. I do have it up and running via ODBC, for what it's worth. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL and Channel Event Logging
On Wed, Oct 13, 2010 at 9:53 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Oct 13, 2010 at 9:31 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Hey all, sorry if this has been covered, but I've not found anything after a couple hours' worth of googling. I can see (and I'm familiar with) all the usual MySQL addon apps once I install Asterisk 1.8.x, but I cannot find any reference to MySQL and the new CEL logging tool other than ODBC. Is this the only method available to use MySQL with CEL at this time? Looking at the CEL config files, I don't see one specifically for MySQL. I do have it up and running via ODBC, for what it's worth. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks mate, I appreciate the reply. That's what I've seen from looking at the configs right now. -- _ -- 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] MySQL and Channel Event Logging
On Wed, Oct 13, 2010 at 9:52 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Wed, Oct 13, 2010 at 9:53 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Oct 13, 2010 at 9:31 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Hey all, sorry if this has been covered, but I've not found anything after a couple hours' worth of googling. I can see (and I'm familiar with) all the usual MySQL addon apps once I install Asterisk 1.8.x, but I cannot find any reference to MySQL and the new CEL logging tool other than ODBC. Is this the only method available to use MySQL with CEL at this time? Looking at the CEL config files, I don't see one specifically for MySQL. I do have it up and running via ODBC, for what it's worth. -- Paul Belanger | dCAP Thanks mate, I appreciate the reply. That's what I've seen from looking at the configs right now. When I wrote the CEL backends, I probably skipped the MySQL stuff, because it would be in the addons stuff for Asterisk. But, if you look at the similarities between the CEL backends, and the CDR backends, you'll probably notice that you could pump out a myseql backend with the same mods in a short amount of time. I would be curious to see how you plan to use that table! Have you mapped out your sql statements yet? murf -- Steve Murphy ParseTree Corp -- _ -- 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] advice re: Page() application
My SIP registration are name sort of like this : phonea-exten1, phone1-exten2, etc. Makes it easy to loop, I can send you a snippet tomorrow. But you have to know in advance all the SIP peer names. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Wednesday, October 13, 2010 10:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] advice re: Page() application Thanks Mike - this does help. The setup will be a local server on the LAN, and hopefully have plenty of snort to handle the load (20-30 phones). I also am not quite ready to put out 1.8 for my users yet. Do you have a snippet of dialplan code you'd be willing to share to loop through a group and grab/build up a list of channels as you describe? That would be enlightening (and probably save me some time)! What I am hearing is - using a second line presence for the Page() function will work; auto-answer should work and I should only page the phones that are not in use. Cassius Original Message Subject: Re: [asterisk-users] advice re: Page() application From: Mike mailto:l...@net-wall.com l...@net-wall.com Date: Thu, October 14, 2010 10:12 am To: 'Asterisk Users Mailing List - Non-Commercial Discussion' mailto:asterisk-users@lists.digium.com asterisk-users@lists.digium.com Hi Cassius, Can`t help for SPA-942, but the Wiki had good info on the Polycoms. Use the Wiki and you`ll do good. Two warnings: It seems to me that the adhoc MeetMe room used by the page application slows things down quite a lot. If you page and have a phone nearby, you`ll hear yourself with quite a bit of delay. It`s very annoying if you`re paging and hearing the page at the same time. Apparently 1.8 supports multicast and will do this differently, but it’ll be a long while before I trust 1.8 to be stable enough for my needs. If you`re doing this over an Internet link (i.e. hosted PBX), keep in mind that because of the MeetMe (I imagine), even if the receiving phones aren’t creating audio, the bandwidth is still is used as if everyone was talking at the same time in a MeetMe room. No biggie if everything is on the LAN, but a bit of a problem if not and you have many phones. And here is a tip: auto-answer is good, but you`ll have to loop through every SIP registration on the phone before using Page() to see if they are being used before adding them to the Page. If not, the phone will not auto-answer (since you`re on a call already) but you`ll have a missed call everytime somebody pages you while you`re on the phone. Users hate that (with reason). You check if each and every phone is being used BEFORE adding them to your page. In other words, if 10 out of 15 phones are idle, Page() only those 10. Besides that, things work as advertised. Mike From: mailto:asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com [ mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Wednesday, October 13, 2010 7:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] advice re: Page() application Hi all, I'm planning a new Asterisk installation; the users want to duplicate the paging function they have with their current Panasonic hybrid system. They dial *3 and announce a held call on line 3, for example, and the announcements comes out of all the desktop phone speakers. I'm planning to implement this using the Page() application in addition to parking the call. The O'Reilly book doesn't talk much about Page(), just says that it dumps the channels into a dynamically created MeetMe room which is quickly torn down. To make this work with typical desktop speakerphones, is there anything I need to do in sip.conf? (I was thinking I might need to set autoanswer=yes, for example). I can use a second line presence on all the phones to support this if necessary; I'm using SPA-942s. I don't want all the phones to ring - just have the announcement audible at each phone without the user needing to pick up. I apologize for not being able to try this out myself - I'm out of the country with no access to sip phones right now. Any help/lessons learned using Page() would be most appreciated! Regards, Cassius Smith _ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users --
