Re: [asterisk-users] Nat=yes
On Thu, Apr 21, 2011 at 5:42 AM, Alexandru Oniciuc alexandru.onic...@trivenet.it wrote: Dear * users, in your opinion, when using a * as a public server, is good practice enabling nat=yes in sip.conf for all the peers? Can anyone imagine a scenario when enabling this parameter (even for peers that don’t require it) can cause problems? Regards and thanks in advance, Alex I asked this same exact question several years ago. There are many replies with different takes. I would skim through Alex's posts, there is really nothing worth reading except it will break the SIP RFC handed down by the internets themselves. I use nat=yes all the time and it works just fine. http://www.mail-archive.com/asterisk-users@lists.digium.com/msg213941.html Nobody actually answered the question about the bad side, they just argued about the SIP RFC. Many others agreed to make it default behavior and that setting nat=yes gives a an extra degree of security. RFCs are great and all, but in the real world, phones just need to work. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar 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] Nat=yes
Hi, When NAT = YES, Asterisk server will extract IP from the network layer. When Nat = No, the Asterisk server will respond to the IP in the SIP header. Am I right? May be such type of options can be helpful for SIP application developers. Can't think of a scenario but If it is set to be YES for all peers, what will happen is that the response to all the SIP request will be routed to the IP in the network layer. IP's in the SIP header will be ignored, should not create a problem. Regards --- On Sun, 4/24/11, Steve Totaro stot...@asteriskhelpdesk.com wrote: From: Steve Totaro stot...@asteriskhelpdesk.com Subject: Re: [asterisk-users] Nat=yes To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, April 24, 2011, 2:13 PM On Thu, Apr 21, 2011 at 5:42 AM, Alexandru Oniciuc alexandru.onic...@trivenet.it wrote: Dear * users, in your opinion, when using a * as a public server, is good practice enabling nat=yes in sip.conf for all the peers? Can anyone imagine a scenario when enabling this parameter (even for peers that don’t require it) can cause problems? Regards and thanks in advance,Alex I asked this same exact question several years ago. There are many replies with different takes. I would skim through Alex's posts, there is really nothing worth reading except it will break the SIP RFC handed down by the internets themselves. I use nat=yes all the time and it works just fine. http://www.mail-archive.com/asterisk-users@lists.digium.com/msg213941.html Nobody actually answered the question about the bad side, they just argued about the SIP RFC. Many others agreed to make it default behavior and that setting nat=yes gives a an extra degree of security. RFCs are great and all, but in the real world, phones just need to work. Thanks, Steve Totaro -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar 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] Nat=yes
On Sun, Apr 24, 2011 at 5:55 AM, Muhammad Ali ali_...@yahoo.com wrote: Hi, When NAT = YES, Asterisk server will extract IP from the network layer. When Nat = No, the Asterisk server will respond to the IP in the SIP header. Am I right? May be such type of options can be helpful for SIP application developers. Can't think of a scenario but If it is set to be YES for all peers, what will happen is that the response to all the SIP request will be routed to the IP in the network layer. IP's in the SIP header will be ignored, should not create a problem. Regards --- On *Sun, 4/24/11, Steve Totaro stot...@asteriskhelpdesk.com* wrote: I am unsure of what you are saying. All I know is that setting nat=yes has never failed me when nat=no has and we are talking countless phones and installs. In the OSI reference model, the Network is layer 3, IP. Call it Network, layer 3, or IP, it is the same. nat=yes breaks the RFC due to NAT but it gets people talking. My customers don't really care for things that don't work. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar 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] Nat=yes
Hi, I am unsure of what you are saying. Just for discussion, if one has a control on the insertion of the IP address in the SIP header, then nat options working can be verified observed. In the OSI reference model, the Network is layer 3, IP. Call it Network, layer 3, or IP, it is the same. All-right, by IP from the network layer I meant, the IP address in the IP Header/Network layer/layer 3. IP from SIP I meant, SIP request generator's IP address in the SIP Header. I missed the word address. My customers don't really care for things that don't work. May be its useful for SIP application developers rather then end customers. Have a good time. Regards --- On Sun, 4/24/11, Steve Totaro stot...@asteriskhelpdesk.com wrote: From: Steve Totaro stot...@asteriskhelpdesk.com Subject: Re: [asterisk-users] Nat=yes To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, April 24, 2011, 3:28 PM On Sun, Apr 24, 2011 at 5:55 AM, Muhammad Ali ali_...@yahoo.com wrote: Hi, When NAT = YES, Asterisk server will extract IP from the network layer. When Nat = No, the Asterisk server will respond to the IP in the SIP header. Am I right? May be such type of options can be helpful for SIP application developers. Can't think of a scenario but If it is set to be YES for all peers, what will happen is that the response to all the SIP request will be routed to the IP in the network layer. IP's in the SIP header will be ignored, should not create a problem. Regards --- On Sun, 4/24/11, Steve Totaro stot...@asteriskhelpdesk.com wrote: I am unsure of what you are saying. All I know is that setting nat=yes has never failed me when nat=no has and we are talking countless phones and installs. In the OSI reference model, the Network is layer 3, IP. Call it Network, layer 3, or IP, it is the same. nat=yes breaks the RFC due to NAT but it gets people talking. My customers don't really care for things that don't work. Thanks, Steve Totaro -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar 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 files
Hello,
Thanks for replying.
Answers below:
On 23 April 2011 18:29, Sherwood McGowan sherwood.mcgo...@gmail.com wrote:
On Sat, Apr 23, 2011 at 11:20 AM, Tiago Geada tiago.ge...@gmail.comwrote:
Hi.
Im having trouble setting variables in channel dialplan and re-using them
in Extension dialplan...
Im using the following call file:
Channel: Local/210332450@ZonNew-Outbound
CallerID: ZonNew-Outbound:49:210332450:
MaxRetries: 5
RetryTime: 10
WaitTime: 60
Account: Outbound210332450
Context: agents
Extension: 888210332450
Set: __PARTNER=ZonNew-Outbound
Set: NUMBER=210332450
-
In Local/210332450@ZonNew-Outbound I Set(bla='blabla');
It seems I cannot re-use this var in extension _888X in context
agents...
Basically the Channel dialplan has a Queue() and in _888X I would
like to know the peer (or interface) that answered it... What can I do?
Thanks in advance
I'm a little confused by It Seems I cannot re-use this var in extension
_888XX in context agentsOf course you can use it...but if you
set bla to a different value in your code where your callfile is processed,
Asterisk will (rightfully so) just set bla = to whatever you set it to
Now, if the callfile doesn't send a channel through the context that
you're trying to set blah, that's a little odd...
Now, as far as retrieving the information about the interface that answered
the calllook in queues.conf.samplethere's a nifty configuration
option:
*setinterfacevar=no ; (the default is no)*
Yes, I am aware of this and I do use it. However, I cannot use
MEMBERINTERFACE variable in dialplan _888X, and that is where I'm
needing it.
Also seems that its two channel legs and the only way would be to use
IMPORT() o SHARED() and for that I would have to know the channel name...
I am right now using IMPORT() like:
Set(CALLERID(num)=${IMPORT(${CHANNEL:0:$[${LEN(${CHANNEL})} -
1]}2,MEMBERNAME)});
but I fee that it is a ugly fix. What if call leg changes from 2 to 3?
That option, when set to yes, causes several variables to be created *just
* prior to the caller being bridged with the queue member...
--
Sherwood McGowan
Telecommunications and VOIP Consultant
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Re: [asterisk-users] DTMF not being sent ( RFC2833 )
I did more testing. Here is a portion of extensions.conf on asterisk-pri: exten = 5,1,Dial(DAHDI/g1/14186939930,30) exten = 6,1,Answer exten = 6,2,Wait(30) exten = 7,1,Dial(DAHDI/g1/14186939930,30,D(132412983#)) Here is an expert from asterisk : exten = 22,1,Dial(SIP/6@pri,30,D(132412983#)) exten = 24,1,Dial(SIP/5@pri,30,D(132412983#)) If I type console dial 24, the DTMFs work poorly, and I see messages like : [Apr 24 11:26:20] DTMF[2691]: channel.c:2907 __ast_read: DTMF end emulation of '1' queued on DAHDI/1-1 [Apr 24 11:26:20] DTMF[2691]: channel.c:2802 __ast_read: DTMF end '1' received on SIP/omnity-0004, duration 60120 ms [Apr 24 11:26:20] DTMF[2691]: channel.c:2842 __ast_read: DTMF end accepted with begin '1' on SIP/omnity-0004 [Apr 24 11:26:20] DTMF[2691]: channel.c:2858 __ast_read: DTMF end passthrough '1' on SIP/omnity-0004 [Apr 24 11:26:20] DTMF[2691]: channel.c:2874 __ast_read: DTMF begin '1' received on DAHDI/1-1 [Apr 24 11:26:20] DTMF[2691]: channel.c:2884 __ast_read: DTMF begin passthrough '1' on DAHDI/1-1 [Apr 24 11:26:20] DTMF[2691]: channel.c:2802 __ast_read: DTMF end '1' received on DAHDI/1-1, duration 39 ms [Apr 24 11:26:20] DTMF[2691]: channel.c:2842 __ast_read: DTMF end accepted with begin '1' on DAHDI/1-1 [Apr 24 11:26:20] DTMF[2691]: channel.c:2851 __ast_read: DTMF end '1' has duration 39 but want minimum 80, emulating on DAHDI/1-1 If I type console dial 22 on asterisk, all the DTMFs are 60ms in length and I get no unusually long DTMFs. If I type console dial 7 on asterisk-pri, all the DTMFs are properly sent, and the remote party sees my DTMFs perfectly. So it would seem that the bug occurs when one asterisk calls the second asterisk which bridges to a DAHDI channel. My next step is too compare the SIP signalling between the two calls. Maybe something is different. What I find really weird is that the DTMF is incorrectly sent from the first asterisk only when the second asterisk bridges to DAHDI. Any ideas? David On 11-04-23 11:48 AM, David wrote: Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from source. Asterisk : spandsp, dahdi, asterisk. Asterisk-pri : spandsp, libpri, dahdi, asterisk wanpipe I eliminated AGI, hard phones, network et al by setting up this extension : exten = 22,1,Dial(SIP/114186939...@pri1.omnity.net,30,D(132412983 mailto:SIP/114186939...@pri1.omnity.net,30,D%28132412983#)) in default. The only other non default setting is in sip.conf I added a outboundproxy ( which does NOT do RTP, only SIP ). I called asterisk from my hard phone ( gxp2000 ) by dialing 22. I see the console DTMF messages indicating the DTMF was sent or received. ( I forgot to keep this output ). I than watch the console DTMF output on asterisk-pri and it showed about half the DTMFs. The pager that was called showed the DTMFs that appeared on the asterisk-pri console. So somewhere between the two machines, the DTMFs have disappeared. So I ran TCPDump on asterisk and saw that close to half of the DTMF events were never sent. tcpdump -i eth0 -n -s 0 dst asterisk-pri-ip -vvv -w ~/dtmf.pcap I imported the file into wireshark on my local machine and confirmed that the dump almost matches what I saw on asterisk-pri. So, problem 1 : Asterisk is not sending all the DTMFs to asterisk-pri. I compared the packet scan to what I saw on asterisk-pri and noticed that between 1 and 3 dtmfs were missing. Problem 2 : Asterisk-pri loses some received DTMFs. I also noticed that some of the DTMFs coming out of asterisk had the wrong Event Duration. I had one DTMF with a duration of about 58000 ( I believe that's 58 seconds ) but I only pressed the button for like 1/3 of a second. What I do not understand is that I in my final test last night was using asterisk 1.6 current with centos ( os that asterisk is developed on from my understanding ) with all default settings ( excluding logger.conf, dialplan and outboundproxy ) and I am having problems with the DTMF. Both servers were installed with CentOS 5.5 and were updated last night, after which I reinstalled asterisk. This did not resolve the issue. I am at wit's end and do not know where to go from here. I would really appreciate it if someone could give me some pointers on where to go next, what additionnal debugging steps I should perform. I would also really appreciate if someone could propose a solution. Please help! David Never give up, never surrender -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar
[asterisk-users] Best modem for chan_datacard
Hi List, I am looking to play around with chan_datacard. Any advice on the best device to test with (that I can find on eBay) ? Regards, Dovid -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar 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] DTMF not being sent ( RFC2833 )
Hello, I traced the SIP packets and saw that the only difference was that the DAHDI channel returns 183 Session progress ( besides the obvious differences such as the To and from tags in sip , session id and rtp ports in the SDP ). I updated my dialplan on asterisk-pri as follows : exten = 6,1,Progress exten = 6,n,Wait(5) exten = 6,n,Answer exten = 6,n,Wait(30) This makes the local channel behave the same as the DAHDI channel. With this in place, the SIP packets for both test calls are identical ( excluding the To header, To Tag, From Tag, SDP ports, SDP session Id and SDP version. Everything else is identical. So the problem appears to be caused in the RTP and not in the SIP. So something about the RTP packets coming from the DAHDI channel on asterisk-pri makes asterisk server send invalid DTMF. David On 11-04-24 11:42 AM, David wrote: I did more testing. Here is a portion of extensions.conf on asterisk-pri: exten = 5,1,Dial(DAHDI/g1/14186939930,30) exten = 6,1,Answer exten = 6,2,Wait(30) exten = 7,1,Dial(DAHDI/g1/14186939930,30,D(132412983#)) Here is an expert from asterisk : exten = 22,1,Dial(SIP/6@pri,30,D(132412983#)) exten = 24,1,Dial(SIP/5@pri,30,D(132412983#)) If I type console dial 24, the DTMFs work poorly, and I see messages like : [Apr 24 11:26:20] DTMF[2691]: channel.c:2907 __ast_read: DTMF end emulation of '1' queued on DAHDI/1-1 [Apr 24 11:26:20] DTMF[2691]: channel.c:2802 __ast_read: DTMF end '1' received on SIP/omnity-0004, duration 60120 ms [Apr 24 11:26:20] DTMF[2691]: channel.c:2842 __ast_read: DTMF end accepted with begin '1' on SIP/omnity-0004 [Apr 24 11:26:20] DTMF[2691]: channel.c:2858 __ast_read: DTMF end passthrough '1' on SIP/omnity-0004 [Apr 24 11:26:20] DTMF[2691]: channel.c:2874 __ast_read: DTMF begin '1' received on DAHDI/1-1 [Apr 24 11:26:20] DTMF[2691]: channel.c:2884 __ast_read: DTMF begin passthrough '1' on DAHDI/1-1 [Apr 24 11:26:20] DTMF[2691]: channel.c:2802 __ast_read: DTMF end '1' received on DAHDI/1-1, duration 39 ms [Apr 24 11:26:20] DTMF[2691]: channel.c:2842 __ast_read: DTMF end accepted with begin '1' on DAHDI/1-1 [Apr 24 11:26:20] DTMF[2691]: channel.c:2851 __ast_read: DTMF end '1' has duration 39 but want minimum 80, emulating on DAHDI/1-1 If I type console dial 22 on asterisk, all the DTMFs are 60ms in length and I get no unusually long DTMFs. If I type console dial 7 on asterisk-pri, all the DTMFs are properly sent, and the remote party sees my DTMFs perfectly. So it would seem that the bug occurs when one asterisk calls the second asterisk which bridges to a DAHDI channel. My next step is too compare the SIP signalling between the two calls. Maybe something is different. What I find really weird is that the DTMF is incorrectly sent from the first asterisk only when the second asterisk bridges to DAHDI. Any ideas? David On 11-04-23 11:48 AM, David wrote: Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from source. Asterisk : spandsp, dahdi, asterisk. Asterisk-pri : spandsp, libpri, dahdi, asterisk wanpipe I eliminated AGI, hard phones, network et al by setting up this extension : exten = 22,1,Dial(SIP/114186939...@pri1.omnity.net,30,D(132412983 mailto:SIP/114186939...@pri1.omnity.net,30,D%28132412983#)) in default. The only other non default setting is in sip.conf I added a outboundproxy ( which does NOT do RTP, only SIP ). I called asterisk from my hard phone ( gxp2000 ) by dialing 22. I see the console DTMF messages indicating the DTMF was sent or received. ( I forgot to keep this output ). I than watch the console DTMF output on asterisk-pri and it showed about half the DTMFs. The pager that was called showed the DTMFs that appeared on the asterisk-pri console. So somewhere between the two machines, the DTMFs have disappeared. So I ran TCPDump on asterisk and saw that close to half of the DTMF events were never sent. tcpdump -i eth0 -n -s 0 dst asterisk-pri-ip -vvv -w ~/dtmf.pcap I imported the file into wireshark on my local machine and confirmed that the dump almost matches what I saw on asterisk-pri. So, problem 1 : Asterisk is not sending all the DTMFs to asterisk-pri. I compared the packet scan to what I saw on asterisk-pri and noticed that between 1 and 3 dtmfs were missing. Problem 2 : Asterisk-pri loses some received DTMFs. I also noticed that some of the DTMFs coming out of asterisk had the wrong Event Duration. I had one DTMF with a duration of about 58000 ( I believe that's 58 seconds ) but I only pressed the button for like 1/3 of a second. What I do not understand is that I in my final test last night was using asterisk
[asterisk-users] DTMF incorrectly sent ( RFC2833 or SIPInfo )
Hello, I will summarize the current situation. I have reduced the bug to two asterisk machines. One of which has a PRI card ( DAHDI channels). The first server calls the second server with SIP. The second server bridges the SIP channel to a DAHDI channel. When I send DTMFs from the first server ( SIP Info or RFC2833) they are incorrectly sent. I setup a second scenario where the first server calls the second with SIP and the second server uses the dialplan to answer, it plays back a message to generate audio. The DTMFs are received perfectly. Asterisk server that is dialing ( first server ): exten = 22,1,Dial(SIP/6...@pri1.omnity.net,30,D(132412983#)) exten = 24,1,Dial(SIP/5...@pri1.omnity.net,30,D(132412983#)) Asterisk server that is answering ( second server ): exten = 5,1,Dial(DAHDI/g1/14186939930,30) exten = 6,1,Progress exten = 6,n,Wait(5) exten = 6,n,Answer exten = 6,n,Playback(vm-loginvm-loginvm-loginvm-loginvm-login) exten = 6,n,Wait(30) Here is the latest console output, notice that the # was sent several times but in reality it was only sent once by the first server. This is the scenario where I dialed 22, the DTMF ( SIP Info ) is sent properly. [Apr 24 12:49:46] DTMF[2844]: channel.c:2907 __ast_read: DTMF end emulation of '1' queued on SIP/omnity-0022 [Apr 24 12:49:46] DTMF[2844]: channel.c:2874 __ast_read: DTMF begin '1' received on SIP/omnity-0022 [Apr 24 12:49:46] DTMF[2844]: channel.c:2878 __ast_read: DTMF begin ignored '1' on SIP/omnity-0022 [Apr 24 12:49:46] DTMF[2844]: channel.c:2802 __ast_read: DTMF end '3' received on SIP/omnity-0022, duration 100 ms [Apr 24 12:49:46] DTMF[2844]: channel.c:2858 __ast_read: DTMF end passthrough '3' on SIP/omnity-0022 [Apr 24 12:49:46] DTMF[2844]: channel.c:2802 __ast_read: DTMF end '2' received on SIP/omnity-0022, duration 100 ms [Apr 24 12:49:46] DTMF[2844]: channel.c:2858 __ast_read: DTMF end passthrough '2' on SIP/omnity-0022 [Apr 24 12:49:46] DTMF[2844]: channel.c:2802 __ast_read: DTMF end '4' received on SIP/omnity-0022, duration 100 ms [Apr 24 12:49:46] DTMF[2844]: channel.c:2858 __ast_read: DTMF end passthrough '4' on SIP/omnity-0022 [Apr 24 12:49:46] DTMF[2844]: channel.c:2802 __ast_read: DTMF end '1' received on SIP/omnity-0022, duration 100 ms [Apr 24 12:49:46] DTMF[2844]: channel.c:2858 __ast_read: DTMF end passthrough '1' on SIP/omnity-0022 [Apr 24 12:49:47] DTMF[2844]: channel.c:2802 __ast_read: DTMF end '2' received on SIP/omnity-0022, duration 100 ms [Apr 24 12:49:47] DTMF[2844]: channel.c:2858 __ast_read: DTMF end passthrough '2' on SIP/omnity-0022 [Apr 24 12:49:47] DTMF[2844]: channel.c:2802 __ast_read: DTMF end '9' received on SIP/omnity-0022, duration 100 ms [Apr 24 12:49:47] DTMF[2844]: channel.c:2858 __ast_read: DTMF end passthrough '9' on SIP/omnity-0022 [Apr 24 12:49:47] DTMF[2844]: channel.c:2802 __ast_read: DTMF end '8' received on SIP/omnity-0022, duration 100 ms [Apr 24 12:49:47] DTMF[2844]: channel.c:2858 __ast_read: DTMF end passthrough '8' on SIP/omnity-0022 [Apr 24 12:49:47] DTMF[2844]: channel.c:2802 __ast_read: DTMF end '3' received on SIP/omnity-0022, duration 100 ms [Apr 24 12:49:47] DTMF[2844]: channel.c:2858 __ast_read: DTMF end passthrough '3' on SIP/omnity-0022 [Apr 24 12:49:48] DTMF[2844]: channel.c:2802 __ast_read: DTMF end '#' received on SIP/omnity-0022, duration 100 ms [Apr 24 12:49:48] DTMF[2844]: channel.c:2858 __ast_read: DTMF end passthrough '#' on SIP/omnity-0022 == Here is the console from the first scenario using SIP Info : [Apr 24 12:50:18] DTMF[2845]: channel.c:2802 __ast_read: DTMF end '1' received on SIP/omnity-0023, duration 100 ms [Apr 24 12:50:18] DTMF[2845]: channel.c:2828 __ast_read: DTMF begin emulation of '1' with duration 100 queued on SIP/omnity-0023 [Apr 24 12:50:18] DTMF[2845]: channel.c:2874 __ast_read: DTMF begin '1' received on DAHDI/1-1 [Apr 24 12:50:18] DTMF[2845]: channel.c:2878 __ast_read: DTMF begin ignored '1' on DAHDI/1-1 [Apr 24 12:50:18] DTMF[2845]: channel.c:2802 __ast_read: DTMF end '1' received on DAHDI/1-1, duration 39 ms [Apr 24 12:50:18] DTMF[2845]: channel.c:2858 __ast_read: DTMF end passthrough '1' on DAHDI/1-1 [Apr 24 12:50:18] DTMF[2845]: channel.c:2907 __ast_read: DTMF end emulation of '1' queued on SIP/omnity-0023 [Apr 24 12:50:18] DTMF[2845]: channel.c:2874 __ast_read: DTMF begin '1' received on DAHDI/1-1 [Apr 24 12:50:18] DTMF[2845]: channel.c:2878 __ast_read: DTMF begin ignored '1' on DAHDI/1-1 [Apr 24 12:50:18] DTMF[2845]: channel.c:2802 __ast_read: DTMF end '1' received on DAHDI/1-1, duration 80 ms [Apr 24 12:50:18] DTMF[2845]: channel.c:2858 __ast_read: DTMF end passthrough '1' on DAHDI/1-1 [Apr 24 12:50:18] DTMF[2845]: channel.c:2802 __ast_read: DTMF end '3' received on SIP/omnity-0023, duration 100 ms [Apr 24 12:50:18] DTMF[2845]: channel.c:2828
[asterisk-users] Realtime and priority labels
In the following example
exten = _1NXXNXX,1,Set(GROUP(outbound)=myprovider)
exten = _1NXXNXX,n,Set(COUNT=${GROUP_COUNT(myprovider@outbound)})
exten = _1NXXNXX,n,NoOp(There are ${COUNT} calls for myprovider)
exten = _1NXXNXX,n,GotoIf($[ ${COUNT} 2 ]?denied : continue)
exten = _1NXXNXX,n(denied),NoOp(There are too many calls up)
exten = _1NXXNXX,n,Hangup()
exten = _1NXXNXX,n(continue),GoSub(callmyprovider,${EXTEN},1)
instead of sequentially numbering the priorities, the n construct is
used. I find that when I attempt this in the realtime extensions table
only, the first priority step is recognized. If I sequentially number
the priorities and add a label, the step is no longer recognized.
Is this behavior by design or an error?
Thanks in advance
Bruce
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[asterisk-users] Repost: Jabber / facebook chat?
On Sunday 17 April 2011, Stefan Gofferje wrote: Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. -S -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar 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] Repost: Jabber / GTalk / hints
On Sunday 17 April 2011, Stefan Gofferje wrote: Hi! Are hints not yet implemented in res_jabber? I have this here: exten = 3000,hint,gtalk/gtalk_account/mari....@gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003@internal : SCCP/6003 State:Unavailable Watchers 0 6002@internal : SCCP/6002 State:IdleWatchers 0 6001@internal : SCCP/6001 State:IdleWatchers 0 6000@internal : SCCP/6000 State:IdleWatchers 0 6004@internal : SIP/sgofferj State:IdleWatchers 0 6200@internal : SCCP/6200 State:Unavailable Watchers 0 3000@internal : gtalk/gtalk_account/ State:Idle Watchers 1 Funnily, the gtalk hint is the only one with a watcher although all hints are hooked in various phones... Any ideas, comments, etc...? -S -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar 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] (no subject)
HI, I am trying to setup a Class 4 termination setup using a kind of channel hunting scenerio. I have some SIP DID numbers assigned from the local telecom provider for termination. MY call comes from my wholesale client and lands on a switch, then it is routed to asterisk. I want asterisk to route this call to my local DID provider on the next available channel with DID number as the new Caller ID. This is just like GSM gateway that recieves the call and then re-originates the call using the next available SIM card number. Can someone help me how can I configure Asterisk to perform this? Thanks Abid. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar 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
