[asterisk-users] [OT] Friday funny
-- Forwarded message -- From: Peter Kunz munged Been there, many, many times. http://xkcd.com/806/ Look at this comic, you will laugh, I guarantee it! Thanks Peter! -- _ -- 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] Does any one uses PortSIP VoIP SDK?
Does it working good with RFC standard? Or where can I get a crack version? 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] Dial plan help
Hi Jigar I am facing issue while generating a dial plan for the following case: all caller should be asked a code to enter than All the callers should be connected one extension. Try DISA component, and then use MeetMe component if you want callers to go to conference or Dial component if you want them to go to extension. I have created a dial plan using vdp I tried submitting it here but I don't know how to extract text version for the same . Visual dialplan outputs standard extensions.conf code. You can get the code by selecting Local deploy option at preferences window or SSH to Asterisk server and check extensions.conf. I was coding dial plans in vi for some time and then switch to Visual Dialplan, much easier and faster, very useful tool. Rayan -- _ -- 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] Cepstral voice quality not good
Actually it is bad only when received on cell phones. Today I listened to the same voices on a Cisco 7942 and they were great. I actually enjoyed listening to them. Not bad on X-Lite either. Previously I was mostly listening to them only through cell phones. So it means it is because of the transcodings at cell phone providers' ends. Bad though because many customers use cell phones exclusively. Maybe if I convert them to gsm format before playing, they'll play better, but will add delay and additional processing because they are converted and played in real time. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-23 6:24 PM, Zeeshan Zakaria zisha...@gmail.com wrote: I am using app_swift. As a side note, demo on their website also generates sounds which at places sounds like robotic. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-23 6:03 PM, Darren Sessions dmsessi...@gmail.com wrote: Are you using app_swift or wav files? On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, I hav... -- _ -- Bandwidth and ... -- _ -- Bandwidth and Colocation P... -- _ -- 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] Dial plan help
I totally agree with Steve's wise advice. One should at least give himself a week learning asterisk fundamentals and related Linux basics before jumping into creating dialplans or setting up Telecom systems. Asterisk's official book's first few chapters cover all the basics which every asterisk user must to know. Otherwise seeking help here won't help because you won't be able to even understand the answers here. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-24 7:59 AM, Rayan Smith rayan.o.sm...@gmail.com wrote: Hi Jigar I am facing issue while generating a dial plan for the following case: all caller should be as... Try DISA component, and then use MeetMe component if you want callers to go to conference or Dial component if you want them to go to extension. I have created a dial plan using vdp I tried submitting it here but I don't know how to extract t... Visual dialplan outputs standard extensions.conf code. You can get the code by selecting Local deploy option at preferences window or SSH to Asterisk server and check extensions.conf. I was coding dial plans in vi for some time and then switch to Visual Dialplan, much easier and faster, very useful tool. Rayan -- _ -- 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] E1 and T1 on the same card, or on the same server
Thanks Kevin to verify this. This would really solve a very big problem for me as E1-T1 conversions has been a big part of my work lately, with no satisfactory and reliable solution yet. I'll propose this card to my client and would love to try it. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-22 6:15 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 10/22/2010 04:05 PM, Zeeshan Zakaria wrote: Hello list, (Resending this email due to a typ... Yes, the cards in question can handle some ports configured as T1 while others are configured as E1. -- 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] Cepstral voice quality
Do you recommend using wav files instead? Will there be any downside of using wav? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocatio... -- _ -- 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] Cepstral voice quality
Well, the downside to wav files is the disk i/o. Asterisk will and does translate the audio frames from ulaw to whatever other codec. Sent from my iPhone On Oct 24, 2010, at 9:42 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Do you recommend using wav files instead? Will there be any downside of using wav? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocatio... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 IAX Registration
Paul, Further to my last, I think I found another small related issue with IAX which is generating the following error: [Oct 24 14:42:12] ERROR[15589]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported To reproduce this issue, setup a phone in iax.conf or your realtime table, goto Zoiper press register in the Preferences / IAX Account. The phone will register correctly when your patch is applied. Then press unregister in Zoiper. On my realtime peers the error then shows up on the console, but for my static iax.conf peers it does not. If you then do a iax2 show peers on the console, the error is displayed. Notice that the value for Host in the command output is a empty string (example below). Name/UsernameHost Mask Port Status 111 (D) 255.255.255.255 0 Unmonitored Initially (for static iax.conf peers) before registration this value is null and does not cause the error to be displayed on a iax2 show peers command (example below). So I'm guessing that somewhere on un-registration this is set to when it should be set to null. Name/UsernameHost Mask Port Status 111 (null) (D) 255.255.255.255 0 Unmonitored Thanks, Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge Sent: 24 October 2010 14:31 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 IAX Registration Paul, I applied your patch to 1.8.0 and I'm happy to report it has fixed the problem I was experiencing. Thanks again. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: 23 October 2010 22:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 IAX Registration On Sat, Oct 23, 2010 at 3:03 PM, Paul Belanger paul.belan...@polybeacon.com wrote: Okay, just reproduced your issue and looking at the code now. :) Ok, think I fixed it. You can either apply this patch to 1.8.0, or svn update the branch I'm working on. Feedback is welcome. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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] Cepstral voice quality not good
I fiddled with the demo version of swift a year or so ago and I had better sound quality if I used the non-8khz versions and had app_swift or asterisk convert it for me (not sure, giving app_swift a regular version seemed to JustWork(tm) On Sat, Oct 23, 2010 at 3:24 PM, Zeeshan Zakaria zisha...@gmail.com wrote: I am using app_swift. As a side note, demo on their website also generates sounds which at places sounds like robotic. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-23 6:03 PM, Darren Sessions dmsessi...@gmail.com wrote: Are you using app_swift or wav files? On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, I hav... -- _ -- 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 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't hear MOH from PSTN
Hello, My setup is : phone - PSTN/ISDN - Patton SN4638 --- Asterisk (Asterisk is in 1.6.1.18, Patton in 5.3) When I call the Asterisk, I can read from console that : - the call comes in, - the line MusicOnHold(,10) in my diaplan is reached and played, - I see RTP packets coming in and out (hundreds of lines such as: Got RTP packet from192.168.102.200:4890 (type 00, seq 005360, ts 2343932047, len 000160) Sent RTP packet to 192.168.102.200:4890 (type 00, seq 036824, ts 082080, len 000160) ) - but I can't hear anything on my end. When I tried this system locally, I could hear the MOH (but now, I can't try any local operation from my location). I was thinking of a codec issue but I can't see why and where. Suggestions ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't hear MOH from PSTN
2010/10/24 Olivier oza_4...@yahoo.fr Hello, My setup is : phone - PSTN/ISDN - Patton SN4638 --- Asterisk (Asterisk is in 1.6.1.18, Patton in 5.3) When I call the Asterisk, I can read from console that : - the call comes in, - the line MusicOnHold(,10) in my diaplan is reached and played, - I see RTP packets coming in and out (hundreds of lines such as: Got RTP packet from192.168.102.200:4890 (type 00, seq 005360, ts 2343932047, len 000160) Sent RTP packet to 192.168.102.200:4890 (type 00, seq 036824, ts 082080, len 000160) ) - but I can't hear anything on my end. When I tried this system locally, I could hear the MOH (but now, I can't try any local operation from my location). I was thinking of a codec issue but I can't see why and where. Suggestions ? Regards PS: If this helps, I'm using default MOH .wav files in /var/lib/asterisk/moh directory. -- _ -- 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] 1 way audio asterisk 1.6
2010/10/21 Zakir Mahomedy z...@mayfair2000.com Hi I wonder if anyone could give some light on SIP NAT. I've having a friken headache with SIP NAT 1 way audio. Client - NAT - NAT - Server Client can hear users from server side but server cant hear client. Ive tried every possible settings externip set localip set NAT= yes / route directmedia yes/ no Ive check the sip headers in the debug mode and its using the external address in both client and server. Ive tried STUn servers etc No luck. any info on this Its for my installation which I am testing. Zakir -- _ -- 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 Which ports are open or forwarded on both firewalls ? Could you post some RTP traces ? -- _ -- 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] ISDN SS7
Hi all, I'm being requested to deploy an IVR service using SS7. I've deployed Asterisk before using ISDN connection, but never with SS7. Can anyone explain me the different between using ISDN and SS7 ? What need I do now to change to use SS7 ?. Many thanks, Giang -- _ -- 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] Default MOH not working on 1.6.1 [SOLVED]
2010/10/24 Olivier oza_4...@yahoo.fr 2010/10/14 Danny Nicholas da...@debsinc.com -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* Thursday, October 14, 2010 3:34 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Default MOH not working on 1.6.1 2010/10/14 Olivier oza_4...@yahoo.fr 2010/10/14 Olivier oza_4...@yahoo.fr Hello, I've configured with the very same script 1 Intel Xeon and 1 Intel Pentium4 machines. On one MOH is working properly On the other, I can read on console, lines such as those bellow but I can't hear anything. In which direction, should I further investigate ? If this help, here is my setup: me ---PSTN-ISDN Patton 4638 ---SIP--- Asterisk 1.6.1.18 -- Started music on hold, class 'default', on SIP/patton-002b == Using SIP RTP CoS mark 5 == Extension Changed 249[subs] new state Ringing for Notify User 749 == Extension Changed 249[subs] new state Ringing for Notify User 750 -- SIP/249-002c is ringing Sent RTP packet to 192.168.102.200:5030 (type 00, seq 043637, ts 000160, len 000160) Sent RTP packet to 192.168.102.200:5030 (type 00, seq 043638, ts 000320, len 000160) Thanks PS: I used the standard i386 Lenny image on the Xeon machine. Should I favor another image, such as amd64 or em64t, instead ? If this matters, I must also add MOH is triggered here by Queue application. I assume MOH is working on Pentium 4 and “failing” on Xeon? Try this snippet Exten = 664,1,answer exten = 664,n,SetMusicOnHold(default) exten = 664,n,WaitMusicOnHold(20) exten = 664,n,Background(vm-goodbye) exten = 664,n,Hangup This should play your default MOH for 20 seconds, then say goodbye and hangup. Hi, I can confirm MOH can't be heard but vm-goodbye file can ! I turned RTP debug on to see what is going on and I can see RTP packets flowing in and out : Got RTP packet from192.168.102.200:4876 (type 08, seq 004857, ts 1590310527, len 000160) Sent RTP packet to 192.168.102.200:4876 (type 08, seq 031524, ts 001440, len 000160) I opened another thread in this list to further detail my setup. A strange thing is that RTP flows seem to use very different timestamps (parameter ts above ?) but comparing with another setup, it doesn't seem to matter. Regards Re-reading you advice, I realized I forgot to type the Answer line. Adding it did it. Thanks for all. It is really strange to realize how applications (Playback, Dial, Queue) have different requirements towards Answer() statement. -- _ -- 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] Can't hear MOH from PSTN [SOLVED]
Adding an Answer() before MusicOnHold made it works. Thanks for everyone that helped ! -- _ -- 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] How to have failover sip interface?
My asterisk machine has 2 nic's. One nic (ETH0) is connected to a cable modem. The other nic (ETH1) is connected to an internal lan. The internal lan also has access to the internet. The cable service, Time-Warner RoadRunner, is great when up, but is not reliable. And sip connections are excellent. The connection through the internal lan (Verizon DSL) is reliable but lousy. Sigh. When the cable is down, the interface connection to the cable modem stays up. An ifconfig shows ETH0 as up. The only way to tell is to ping an outside address. I thought of bonding. But that won't work since it will see ETH0 as up, even if the cable service is down. Is there a way to implement network failover that actually checks for true internet connection? This way I can keep my sip connection up, even if degraded. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN SS7
Hi cary, Can you recommend me what add-on vendors I should use ? Can a open source solution such as chan_ss7 or libss7 support many conncurrent calls (for example 240 calls) ? Thanks From: Cary Fitch ca...@usawide.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sun, October 24, 2010 9:33:28 AM Subject: Re: [asterisk-users] ISDN SS7 SS7 is an inter-telco system using a separate network for all signaling. You must have an SS7 network connection before anything will work. Then the T1 Spans run 24 64k audio paths. The SS7 net exchanges the call data and connection info between the switches. Asterisk doesn’t support SS7 natively although I believe there are one or more add-on vendors. Cary Fitch From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang Sent: Sunday, October 24, 2010 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ISDN SS7 Hi all, I'm being requested to deploy an IVR service using SS7. I've deployed Asterisk before using ISDN connection, but never with SS7. Can anyone explain me the different between using ISDN and SS7 ? What need I do now to change to use SS7 ?. Many thanks, Giang -- _ -- 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] ISDN SS7
I do not have knowledge of the SS7 vendors for Asterisk. Using redundant 56k data channels, we handle calls via 6 DS3s (672 X 6 calls) from the PSTN on a commercial telephone switch, with no issues at all. SS7 can support any number of simultaneous calls depending only on the bandwidth of the SS7 channels. SS7 is always done on a redundant channel basis since it is so important. Cary _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang Sent: Sunday, October 24, 2010 12:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ISDN SS7 Hi cary, Can you recommend me what add-on vendors I should use ? Can a open source solution such as chan_ss7 or libss7 support many conncurrent calls (for example 240 calls) ? Thanks _ From: Cary Fitch ca...@usawide.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sun, October 24, 2010 9:33:28 AM Subject: Re: [asterisk-users] ISDN SS7 SS7 is an inter-telco system using a separate network for all signaling. You must have an SS7 network connection before anything will work. Then the T1 Spans run 24 64k audio paths. The SS7 net exchanges the call data and connection info between the switches. Asterisk doesn't support SS7 natively although I believe there are one or more add-on vendors. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang Sent: Sunday, October 24, 2010 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ISDN SS7 Hi all, I'm being requested to deploy an IVR service using SS7. I've deployed Asterisk before using ISDN connection, but never with SS7. Can anyone explain me the different between using ISDN and SS7 ? What need I do now to change to use SS7 ?. Many thanks, Giang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 IAX Registration
On Sun, Oct 24, 2010 at 10:06 AM, Nic Colledge n...@njcolledge.net wrote: Further to my last, I think I found another small related issue with IAX which is generating the following error: Do you mind collecting a debug log [1]? Having some issues reproducing this. [1] http://svn.asterisk.org/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 IAX Registration
Paul, I made a debug log of the register and unregister process for a single Zoiper client using IAX and have emailed it direct to you. The error shows in the file as: [Oct 24 19:07:32] ERROR[1403] netsock2.c: getnameinfo(): ai_family not supported Thanks, Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: 24 October 2010 18:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 IAX Registration On Sun, Oct 24, 2010 at 10:06 AM, Nic Colledge n...@njcolledge.net wrote: Further to my last, I think I found another small related issue with IAX which is generating the following error: Do you mind collecting a debug log [1]? Having some issues reproducing this. [1] http://svn.asterisk.org/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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] baffled by defaultuser on aastra 9133i
1.6.2.13, sip.conf: [155] type=friend context=longdistance callerid=Admin 155 secret=test host=dynamic dtmfmode=rfc2833 allow=all defaultuser=155-trust On aastra: Basic SIP Authentication Settings Screen Name Phone Number 155 Caller ID 155 Authentication Name 155-trust Password test But: WARNING[1737]: chan_sip.c:12800 check_auth: username mismatch, have 155, digest has 155-trust NOTICE[1737]: chan_sip.c:21687 handle_request_register: Registration from '155 sip:1...@10' failed for '10..' - Username/auth name mismatch How do I set this so the Authenication is not the same as the extension? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chan variables for peer
Hi all,
I used to configure each of my sip clients with a unique identifier via
setvar. These clients were all configured as friends.
However, now that I've got some Polycom phones, which MUST be peers, I am
unable to define this variable.
For example, this works:
[friend-client]
context = default
accountcode = pcc
type = friend
username = username
secret = ya,right
host = dynamic
nat = yes
canreinvite = no
callerid = User 155512345
mailbox = 155512...@customers,123456
setvar = id=123
However, if I change the type to peer, I am unable to get a value for ${id}.
Is this a known limitation, or am I doing something wrong? If this won't
work, is there a work-around?
--
Take care and have fun,
Mike Diehl.
--
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[asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice
Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? 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] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese rsre...@gmail.com wrote: Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? I wrote one last week: http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/ Also: http://www.davidvossel.com/?p=28 -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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] E1 configuration
Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5-- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,t) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (local, 21341400, 2) exited non-zero on 'SIP/4804-' The boad has succesfully installed: Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) the channels are correct and mfcr2 too, but the calls dont go out. Thanks for any 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] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Sun, Oct 24, 2010 at 7:06 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese rsre...@gmail.com wrote: Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? I wrote one last week: http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/ Also: http://www.davidvossel.com/?p=28 Paul, It seems you were using a beta/SVN release for your example. Do the following two packages need to be installed if using the stable 1.6.0 release before building from source? I ask as I am unable to dial out. $ apt-get install libikesemel-dev $ apt-get install libssl-dev Secondly, do you know if the username/password are sent in clear text to the Google? Thanks, Stephen -- _ -- 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] E1 configuration
Forget it !! After several attempts, I have solved !!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 24 Oct 2010 22:28:16 -0200 Subject: [asterisk-users] E1 configuration Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5-- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,t) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (local, 21341400, 2) exited non-zero on 'SIP/4804-' The boad has succesfully installed: Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) the channels are correct and mfcr2 too, but the calls dont go out. Thanks for any 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 -- _ -- 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] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Sun, Oct 24, 2010 at 9:24 PM, Stephen Reese rsre...@gmail.com wrote: On Sun, Oct 24, 2010 at 7:06 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese rsre...@gmail.com wrote: Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? I wrote one last week: http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/ Also: http://www.davidvossel.com/?p=28 Paul, It seems you were using a beta/SVN release for your example. Do the following two packages need to be installed if using the stable 1.6.0 release before building from source? I ask as I am unable to dial out. $ apt-get install libikesemel-dev $ apt-get install libssl-dev Secondly, do you know if the username/password are sent in clear text to the Google? I installed the two packages previously mentioned but still lack outbound dialing. I enabled debugging and am getting the following messages. I double checked the password and even changed it to one without special characters but still the same results. JABBER: gmail INCOMING: failure xmlns=urn:ietf:params:xml:ns:xmpp-saslnot-authorized//failure [Oct 24 23:07:55] ERROR[28785]: res_jabber.c:1693 aji_act_hook: JABBER: encryption failure. possible bad password. JABBER: gmail INCOMING: /stream:stream [Oct 24 23:07:55] ERROR[28785]: res_jabber.c:1576 aji_act_hook: aji_act_hook was called with out a packet [Oct 24 23:07:55] WARNING[28785]: res_jabber.c:1391 aji_recv: Parsing failure: Hook returned an error. [Oct 24 23:07:55] WARNING[28785]: res_jabber.c:2742 aji_recv_loop: JABBER: Got hook event. [Oct 24 23:07:55] WARNING[28785]: res_jabber.c:2753 aji_recv_loop: JABBER: socket read error -- _ -- 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] Best way to recording the hold time for a Queue agent or an extension
Anything on this guys? I am sure someone had the need to record the HOLD time or maybe it is already being recorded somewhere? Any thoughts are appreciated. Thanks, Bruce On Wed, Oct 20, 2010 at 3:30 AM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, We are using Queuemetrics but it doesn't Record the Hold Time as it's never logged on the queue_log file. However, when an agent or an extension presses HOLD button on their phone, asterisk does create an event for Music On Hold which is logged in the /var/log/asterisk/full. I want to record the total hold time for an extension and save it with an epoch time stamp. What is the best approach to this? read and parse /var/log/asterisk/full in a cron job every few seconds? Have a presistent PHP-AGI connection to check for hold time events? As much detail as possible on above approaches or other ideas are most appreciated. 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] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Sunday, October 24, 2010 05:23:13 pm Stephen Reese wrote:
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
For Google Voice, I use an ipKall number for the inbound trunk. Here are the
relevant sections of my extensions.conf:
; inbound ipKall trunk (to which Google Voice makes the connection)
[ipkall]
exten = ipKall-number,1,GotoIf($[${DB_EXISTS(gv/channel)} = 1]?gv)
same = n,Goto(default,s,1)
same = n(gv),Bridge(${DB_DELETE(gv/channel)})
same = n,AGI(gv/gv.agi,hangup)
same = n,Hangup()
; outbound Google Voice initiation
[gv-out]
exten = _X.,1,AGI(gv/gv.agi,call)
same = n,While($[${DB_EXISTS(gv/channel)} = 1])
same = n,Wait(0.3)
same = n,EndWhile()
same = n,Hangup()
And the AGI (written in Bash) is here:
http://messinet.com/trac/wiki/AsteriskGVGateway
http://messinet.com/trac/browser/gv/gv.agi
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
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[asterisk-users] xpp_fxloader fails to load Astribank firmware on Ubuntu Lucid
I am running Asterisk on Ubuntu 2.6.32-25-server with asterisk 1.6.2.5-0ubuntu1 and dahdi 2.2.1-0ubuntu2. The machine has a passive HCF-based PCI ISDN card and an Astribank 8 attached. The ISDN card works fine. r...@servaction:~# lsusb Bus 001 Device 002: ID 04b4:8613 Cypress Semiconductor Corp. CY7C68013 EZ-USB FX2 USB 2.0 Development Kit r...@servaction:/usr/share/dahdi# ./xpp_fxloader usb 'xpp_fxloader'[1416]: - FIRMWARE LOADING: (usb) [0 devices] Got all 0 devices 'xpp_fxloader'[1446]: - FIRMWARE IS LOADED Needless to say, 0 devices have the Astribank firmware loaded and lsusb output is the same. I tried loading with the raw fxload command: fxload -t fx2 -D /dev/bus/usb/001/002 -I /usr/share/dahdi/USB_FW.hex Note that /proc/bus/usb is no longer mounted in this version of Ubuntu, so I loaded to /dev/bus/usb. lsusb output no shows: Bus 001 Device 002: ID :ff00 Obviously incorrect and Astribank not detected with dahdi_hardware. I bought the Astribank 8 years ago when they first came out, so it is one of the first, but was working back then. I also downloaded and compiled the latest version of dahdi - no difference in outcome. Thanks in advance for any help. David -- _ -- 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
