[asterisk-users] Asterisk OUtbound IVR Recording
HI, I have a scenario like the following . A user clicks on the web page . This triggers an outbound call to users phone number . Now the user has to leave a message . What is the best way of doing this ? Do we have any example of such a dial plan . Regards Mahesh -- _ -- 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] SPA-2102 sending local IP instead of WAN IP in SIP packets
On 10/08/2010 10:16 PM, bruce bruce wrote: I said previously, Asterisk receives packets like extens...@192.168.0.10 mailto:extens...@192.168.0.10 is trying to register to it. So, Asterisk sends out to local LAN an ACK which obviously is not right. SPA-2102 should send SIP request like extens...@123.123.123.123 mailto:extens...@123.123.123.123 (public IP). If you set 'nat=yes' in the sip.conf peer entry for that device in Asterisk, Asterisk will reply to the IP address and port number the REGISTER request was received from, not the address in the Contact header provided in the request itself. It will also record this address and port number as the location of that peer for future INVITE messages to be sent to it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for a better ATA
On 10/09/2010 06:36 AM, Jeff LaCoursiere wrote: On Fri, 8 Oct 2010, Bryant Zimmerman wrote: I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF talkoff and T38 Any advise before I start testing with these brands would be apperciated. Any better option you may know of. Thanks for any input Bryant I'm curious which of the above ills you attribute to the Linksys (assuming an SPA2102? The PAP2T does have the T38 problem I believe). Its basically the defacto standard for all the giant ITSPs. Perhaps your problem is one that could be rectified in some way. I have also tried Grandstream and Audiocodes (still use the MP-124s in certain situations) and have found that the SPA2102s work the best for us... The PAP2 and PAP2T do not support T.38. The SPA2102 and SPA3102 support it, but have a number of quirks. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice quality assessment in Asterisk
On Fri, Oct 08, 2010 at 05:51:48PM +0400, Sevana Oy wrote: One quick clarification please... With Fluke ACEs you measure MOS according G.107, E-model, right? Don't know if it is g.107 but it has to be something similar if it isn't. I'm personally more interessted in the network statitics. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-2102 sending local IP instead of WAN IP in SIP packets
And that is exactly what is done on the device: Nat=yes but Asterisk still sees the SIP packet coming in to register with a local IP an so it responds to a local IP which doesn't even exist on the Asterisk network. This is what frustrates me that it's not so straight forward to Asterisk to obtain the proper public IP of the device from the IP packet headers rather than the SIP packets. Thanks On Sat, Oct 9, 2010 at 8:27 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 10/08/2010 10:16 PM, bruce bruce wrote: I said previously, Asterisk receives packets like extens...@192.168.0.10 mailto:extens...@192.168.0.10 is trying to register to it. So, Asterisk sends out to local LAN an ACK which obviously is not right. SPA-2102 should send SIP request like extens...@123.123.123.123 mailto:extens...@123.123.123.123 (public IP). If you set 'nat=yes' in the sip.conf peer entry for that device in Asterisk, Asterisk will reply to the IP address and port number the REGISTER request was received from, not the address in the Contact header provided in the request itself. It will also record this address and port number as the location of that peer for future INVITE messages to be sent to it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-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] SPA-2102 sending local IP instead of WAN IP in SIP packets
Am 09.10.2010 20:34, schrieb bruce bruce: And that is exactly what is done on the device: Nat=yes but Asterisk still sees the SIP packet coming in to register with a local IP an so it responds to a local IP which doesn't even exist on the Asterisk network. This is what frustrates me that it's not so straight forward to Asterisk to obtain the proper public IP of the device from the IP packet headers rather than the SIP packets. Thanks when you do a sip debug do you see something like this: SIP read from 192.168.0.2 for example or do you see the internal ip only in the contact header? if its only in the contact header everything is ok. if not you maybe have a network problem like SIP ALG on your router. Asterisk can only work with the data which are received. turning NAT on or off only switch between the IP in the contact header or the source IP but if your device in between like a router does something wrong like faking packet source ips asterisk cant fix this. i dont know what kind of router you use, but have a look at SIP ALG and turn this off, if possible. best regards stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk OUtbound IVR Recording
cmd record ? On Sat, Oct 9, 2010 at 1:28 AM, Govind, Mahesh (NSN - IN/Bangalore) mahesh.gov...@nsn.com wrote: HI, I have a scenario like the following . A user clicks on the web page . This triggers an outbound call to users phone number . Now the user has to leave a message . What is the best way of doing this ? Do we have any example of such a dial plan . Regards Mahesh -- _ -- 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] looking for a better ATA
Hi, On 10/08/2010 11:28 PM, Bryant Zimmerman wrote: I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF talkoff and T38 Any advise before I start testing with these brands would be apperciated. Any better option you may know of. I take it that a pci or usb fxo interface is not what you are looking for - or doesn't fit your requirements? Sebastian Thanks for any input 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] REINVITE with Auth Credentials has different SDP Codec
What am I doing wrong...to get no responses at all Thx From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo Sent: Thursday, October 07, 2010 7:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] REINVITE with Auth Credentials has different SDP Codec Hi I have a call from Service Provider (SP) to Asterisk to User User sends a T38 REINVITE Asterisk passes that to SP SP challenges the INVITE Asterisk sends INVITE with credentials but sends G711ulaw in the SDP instead of T38 udptl... Obviously Fax fails.. Any ideas on how I can maintain the T38 SDP when SP challenges the mid-Call T38 REINVITE? -- _ -- 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] Dahdi missing
Hi, Trying to configure my tdm410p card, my dahdi in asterisk cli was missing. !ael agentagi cdr channel cli config console core database devstate dialplan dnsmgr dundifeatures file group hangup help http iax2 indication keys locallogger manager meetme mgcp minivm mixmonitor module moh no originateparkedcalls phoneprovpri queuerealtime reload rtcp rtp say sip skinny sla stun timing transcoder udptlulimit unistim voicemail Anybody know what file control the presence or not od dahdi in asterisk cli? 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] Dahdi missing
On 10/9/10 11:54 PM, Flavio Miranda wrote: Trying to configure my tdm410p card, my dahdi in asterisk cli was missing. [snip] Anybody know what file control the presence or not od dahdi in asterisk cli? I don't know what version of Asterisk you're running, but I'll take a guess and say the output of module load chan_dahdi.so on the Asterisk CLI will result in an error that will give you the clue you need. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi missing
Hi Shaun, Thanks! Actully, my system.conf in /etc/dahdi was wrong or at least , imcomplete. I am trying for a long time to configure one TDM 410p board and one TE110p om the server . so far without success! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 10 Oct 2010 00:04:27 -0500 From: sruff...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dahdi missing On 10/9/10 11:54 PM, Flavio Miranda wrote: Trying to configure my tdm410p card, my dahdi in asterisk cli was missing. [snip] Anybody know what file control the presence or not od dahdi in asterisk cli? I don't know what version of Asterisk you're running, but I'll take a guess and say the output of module load chan_dahdi.so on the Asterisk CLI will result in an error that will give you the clue you need. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-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
