Re: [asterisk-users] Update problem | CLI commands missing
Hi List, is there somebody how is able to help me here? Or at least to get more details why this occurs? best regards Christoph Am 08.06.2011 18:00, schrieb Christoph Timm: Hi List, I'm running into trouble, if I perform a 'yum update' on my AsteriskNOW. Currently I'm running Asterisk 1.8.3.3. I have the following problem, if I do the update to the actual 1.8.4.2. There are several commands on the CLI which are not working or even not present like core show uptime (not working) core restart (not present) core show version (not present) my Skype for Asterisk is also not loaded correctly. 190 modules are loaded, if I do a 'module show'. I miss also some messages in the log like [Jun 7 21:21:31] VERBOSE[3449] loader.c: func_version.so = (Get Asterisk Version/Build Info). Does anyone know something about this problem? best regards Christoph -- _ -- 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] Connected Line ID
Doug, I see that this patch is for 1.6.0.1 But we use version 1.6.2.12. And if I can see it, this patch is already included in version 1.6.2.12. Or am I wrong? Regards, Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle Verzonden: 10-06-2011 14:01 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Connected Line ID Arjan Kroon | Mobillion wrote: But are there also pathes for version 1.6 The last patch available for the 1.6 series was for 1.6.0.1: https://issues.asterisk.org/jira/browse/8824 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] ITSP failover for PRI
2011/6/19 Claude Hayn chayn...@gmail.com ITSP failover for PRI ** ** Hello All, ** ** We’re using an Asterisk based SIP-T1 trunking gateway and would like to implement failover between two ITSPs. ** ** What about incoming calls ? Do you have a way to have calls that normally comes from ITPS1 to comes from ITSP2 ? -- _ -- 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] ITSP failover for PRI
On 06/20/2011 04:20 AM, Olivier wrote: What about incoming calls ? Do you have a way to have calls that normally comes from ITPS1 to comes from ITSP2 ? No, there is no BGP for the PSTN. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] SMS with Asterisk
Two requests, not from me but the community. 1. Don't top post 2. When you find your solution, reply to this thread so others will be (silver) spoon fed the answers and blindly accept them without trying things and going through a learning curve and experimentation when they find your post in Google. Thanks, Steve T On Mon, Jun 20, 2011 at 1:44 AM, virendra bhati virbh...@gmail.com wrote: Hi Steve, Thanks for share your knowledge. I will revert back to you after testing with asterisk. On Sun, Jun 19, 2011 at 6:46 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Sun, Jun 19, 2011 at 8:49 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Sun, Jun 19, 2011 at 5:13 AM, virendra bhati virbh...@gmail.com wrote: Hi List, I have installed Kannel server into my Linux server. I have asterisk installed into the same server. Now I want to connect both opensource project. As per the VoIP-info website I read that in asterisk there is an option to send SMS. You how to do it. If you have any idea then please help me so thatI will make asterisk as per my need. - Thanks and regards Virendra Bhati +91-9172341457 virbh...@gmail.com Software Engineer Asterisk has some built in features for SMS. You don't need them, you have already setup Kennal which is light years ahead of Asterisk's native in SMS apps and features. The way I do it is to use the System application. It allows you to run programs and such. With system, I call a program called Lynx which is just a simple text web browser.) to open hit a URL that Kannel deciphers and sends the SMS however you Kannel setup. The URL contains all of the information needed to send the SMS, so part of the URL is the destination phone number, part is the body, obviously you need to set your variables in Asterisk and then use the variables in in the Lynx URL. I just found this article that should answer most if not all of your questions. It work just fine for me at many locations. http://www.kannel.org/download/1.4.1/userguide-1.4.1/userguide.html#AEN834 Thanks, Steve Totaro This link show how to send SMS using HTTP(s) and the format of the URL. http://www.kannel.org/download/1.4.1/userguide-1.4.1/userguide.html#AEN4201 The previous link is good news to me. Now I can do anything by hitting a URL. it is so simple. Thanks Steve T -- _ -- 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 and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- 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] ITSP failover for PRI
2011/6/20 Alex Balashov abalas...@evaristesys.com On 06/20/2011 04:20 AM, Olivier wrote: What about incoming calls ? Do you have a way to have calls that normally comes from ITPS1 to comes from ITSP2 ? No, there is no BGP for the PSTN. Yes, that's what I thought but you never know ;-) (Maybe SS7 offers such redundancy but I've got no experience of any king in this domain). -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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-**usershttp://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] ITSP failover for PRI
On 06/20/2011 05:13 AM, Olivier wrote: Yes, that's what I thought but you never know ;-) (Maybe SS7 offers such redundancy but I've got no experience of any king in this domain). SS7 certainly offers link redundancy, but the issue is that your numbers can't just be flash-ported to a different underlying carrier. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] Connected Line ID
Arjan Kroon | Mobillion wrote: And if I can see it, this patch is already included in version 1.6.2.12. Or am I wrong? That I can't answer. I'm still using 1.4.x and am experimenting with 1.8.x. I recall reading that it wasn't supported directly until 1.8 without patches. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Connected Line ID
Oke, But is there a patch from version 1.6.2.12? Greeting, Arjan -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle Verzonden: 20-06-2011 11:36 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Connected Line ID Arjan Kroon | Mobillion wrote: And if I can see it, this patch is already included in version 1.6.2.12. Or am I wrong? That I can't answer. I'm still using 1.4.x and am experimenting with 1.8.x. I recall reading that it wasn't supported directly until 1.8 without patches. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Connected Line ID
It is not included. It was supposed to be included in 1.6.3, but that verison of Asterisk was never released, it became 1.8. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon|Mobillion Sent: Monday, June 20, 2011 4:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Connected Line ID Doug, I see that this patch is for 1.6.0.1 But we use version 1.6.2.12. And if I can see it, this patch is already included in version 1.6.2.12. Or am I wrong? Regards, Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle Verzonden: 10-06-2011 14:01 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Connected Line ID Arjan Kroon | Mobillion wrote: But are there also pathes for version 1.6 The last patch available for the 1.6 series was for 1.6.0.1: https://issues.asterisk.org/jira/browse/8824 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Cisco IP Phones and Skinny in asterisk
Dears; OK, I have two things now: 1) When I do reload from the asterisk CLI, then all the skinny phones are reset. This is very bad thing, how to avoid this from happening in each reload? Even if the reload will be done to take sip configuration !! 2) The line tone that is heared (the normal too tone which is heared when picking up the handset to place a call), now: while dialing the digits, I stay hear the tooo !!! It start ringing at the destination and I am still hearing the too, the destination answer the call and I am still hearing the t. How to resolve this? Please note that currently I am not giving the Phone any files from the TFTP, I just give the Phone the TFTP IP address (which takes it from the DHCP option) and it come to asterisk and register. I am able to call the extension of this Phone and it rings, but when pickup the handset to answer the call, I just hear t even the caller with me in the line and he is saying Hellooo but at Cisco Skinny Phone, I do not hear his voice, I just hear the to. Appreciate the kindly help. Regards Bilal --- The Asterisk version is 1.8.3.2 The Cisco IP Phone is 7942G and it is running now skinny. The used TFTP is tftp-server which is installed in fedora. I placed the following two files (but look like it was not taken from the TFTP, as nothing appeared in the messages), but I am able to to ping from the asterisk box to the vlan that the Phone is connected, so no problem in the reachability: SEPB8BEBF22AB62.cnf.xml xmlDefault.CNF.XML Are the files name correct? Or the Cisco IP Phone 7942G are not working fine with Asterisk or the tftp-server? Cisco has changed the file name format a few times, so you may want to copy xmlDefault.CNF.XML to XMLDefault.cnf.xml The more important steps is how have you configured the phone to locate the TFTP server? Are you using option 150 in DHCP, or manually setting the TFTP server address on the phone. Technically you do not need a TFTP server, since the Skinny phones will try to use the TFTP server address for registration, so you can just set that address to point to your asterisk server. A TFTP server is needed if you want custom ringtones or to manage software updates. For small setups or my home, I skipped setting up the TFTP server until I wanted to update the phone firmware. 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] SMS with Asterisk
But Steve... didn't you just top post? On Mon, Jun 20, 2011 at 10:52 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: Two requests, not from me but the community. 1. Don't top post 2. When you find your solution, reply to this thread so others will be (silver) spoon fed the answers and blindly accept them without trying things and going through a learning curve and experimentation when they find your post in Google. Thanks, Steve T On Mon, Jun 20, 2011 at 1:44 AM, virendra bhati virbh...@gmail.com wrote: Hi Steve, Thanks for share your knowledge. I will revert back to you after testing with asterisk. On Sun, Jun 19, 2011 at 6:46 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Sun, Jun 19, 2011 at 8:49 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Sun, Jun 19, 2011 at 5:13 AM, virendra bhati virbh...@gmail.com wrote: Hi List, I have installed Kannel server into my Linux server. I have asterisk installed into the same server. Now I want to connect both opensource project. As per the VoIP-info website I read that in asterisk there is an option to send SMS. You how to do it. If you have any idea then please help me so thatI will make asterisk as per my need. - Thanks and regards Virendra Bhati +91-9172341457 virbh...@gmail.com Software Engineer Asterisk has some built in features for SMS. You don't need them, you have already setup Kennal which is light years ahead of Asterisk's native in SMS apps and features. The way I do it is to use the System application. It allows you to run programs and such. With system, I call a program called Lynx which is just a simple text web browser.) to open hit a URL that Kannel deciphers and sends the SMS however you Kannel setup. The URL contains all of the information needed to send the SMS, so part of the URL is the destination phone number, part is the body, obviously you need to set your variables in Asterisk and then use the variables in in the Lynx URL. I just found this article that should answer most if not all of your questions. It work just fine for me at many locations. http://www.kannel.org/download/1.4.1/userguide-1.4.1/userguide.html#AEN834 Thanks, Steve Totaro This link show how to send SMS using HTTP(s) and the format of the URL. http://www.kannel.org/download/1.4.1/userguide-1.4.1/userguide.html#AEN4201 The previous link is good news to me. Now I can do anything by hitting a URL. it is so simple. Thanks Steve T -- _ -- 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 and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- 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] different format in asterisk
Hi In asterisk channel ,I so number of variable regarding the Codec ,Can anyone explain what are those variable variable means.Below are the variables 1. chan-readformat 2. chan-writeformat 3. chan -rawreadformat 4. chan -rawwriteformat 5. chan-nativeformats Thanks Nikhil -- _ -- 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] Connected Line ID
Arjan Kroon | Mobillion wrote: But is there a patch from version 1.6.2.12? Not that I can see. You could try applying the patches against that version and see if they apply cleanly. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Connected Line ID
On Mon, Jun 20, 2011 at 5:39 AM, Arjan Kroon | Mobillion arjan.kr...@mobillion.nl wrote: Oke, But is there a patch from version 1.6.2.12? Greeting, Arjan -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle Verzonden: 20-06-2011 11:36 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Connected Line ID Arjan Kroon | Mobillion wrote: And if I can see it, this patch is already included in version 1.6.2.12. Or am I wrong? That I can't answer. I'm still using 1.4.x and am experimenting with 1.8.x. I recall reading that it wasn't supported directly until 1.8 without patches. Doug I am using 1.8 now, but I had updated the patch for SIPCalledRPID() for 1.6.2 and was using it successfully. http://pastebin.com/K1mmGU1c Ryan -- _ -- 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] Connected Line ID
We try the patch against the version 1.6.12 but we got many reject failures. (see below) Unfortunately we have to upgrade to 1.8 Tx, Arjan File to patch: apps/app_dial.c patching file apps/app_dial.c Hunk #1 FAILED at 111. Hunk #2 FAILED at 123. Hunk #3 FAILED at 164. Hunk #4 succeeded at 466 (offset 234 lines). Hunk #5 FAILED at 497. Hunk #6 FAILED at 512. Hunk #7 FAILED at 524. Hunk #8 FAILED at 539. Hunk #9 succeeded at 435 (offset 12 lines). Hunk #10 FAILED at 510. Hunk #11 succeeded at 767 (offset 247 lines). Hunk #12 succeeded at 560 (offset 15 lines). Hunk #13 succeeded at 854 (offset 247 lines). Hunk #14 succeeded at 662 (offset 15 lines). Hunk #15 succeeded at 962 (offset 249 lines). Hunk #16 succeeded at 850 with fuzz 2 (offset 25 lines). Hunk #17 succeeded at 1394 (offset 260 lines). Hunk #18 succeeded at 1263 with fuzz 1 (offset 42 lines). Hunk #19 succeeded at 1572 (offset 274 lines). Hunk #20 FAILED at 1597. Hunk #21 succeeded at 1400 (offset 25 lines). Hunk #22 FAILED at 1490. Hunk #23 succeeded at 1817 (offset 260 lines). 10 out of 23 hunks FAILED -- saving rejects to file apps/app_dial.c.rej patching file apps/app_directed_pickup.c Hunk #1 succeeded at 94 (offset 36 lines). patching file apps/app_followme.c Hunk #1 succeeded at 832 with fuzz 2 (offset 37 lines). patching file apps/app_queue.c Hunk #1 succeeded at 820 (offset 306 lines). Hunk #2 FAILED at 2487. Hunk #3 FAILED at 2874. 2 out of 3 hunks FAILED -- saving rejects to file apps/app_queue.c.rej patching file channels/chan_agent.c Hunk #1 succeeded at 796 (offset 96 lines). patching file channels/chan_dahdi.c Hunk #1 succeeded at 3164 (offset 1033 lines). Hunk #3 succeeded at 3222 with fuzz 2 (offset 1032 lines). Hunk #4 FAILED at 3289. Hunk #5 succeeded at 2404 with fuzz 1 (offset 27 lines). Hunk #6 succeeded at 3458 (offset 1032 lines). Hunk #7 FAILED at 3539. Hunk #8 succeeded at 2791 (offset 76 lines). Hunk #9 succeeded at 3762 (offset 1033 lines). 2 out of 9 hunks FAILED -- saving rejects to file channels/chan_dahdi.c.rej patching file channels/chan_h323.c Hunk #1 succeeded at 609 (offset -2 lines). patching file channels/chan_iax2.c Hunk #1 FAILED at 279. Hunk #2 succeeded at 4340 (offset 1033 lines). Hunk #3 FAILED at 4385. Hunk #5 succeeded at 4849 (offset 1270 lines). Hunk #7 succeeded at 5354 (offset 1365 lines). Hunk #8 succeeded at 4015 (offset -8 lines). Hunk #9 succeeded at 7417 (offset 1479 lines). Hunk #10 succeeded at 8033 (offset 63 lines). Hunk #11 succeeded at 11047 (offset 1602 lines). Hunk #12 FAILED at 11577. Hunk #13 FAILED at 12079. Hunk #14 FAILED at 12320. 5 out of 14 hunks FAILED -- saving rejects to file channels/chan_iax2.c.rej patching file channels/chan_local.c Hunk #1 FAILED at 473. 1 out of 1 hunk FAILED -- saving rejects to file channels/chan_local.c.rej patching file channels/chan_mgcp.c Hunk #1 succeeded at 916 (offset 4 lines). patching file channels/chan_phone.c patching file channels/chan_sip.c Hunk #1 FAILED at 988. Hunk #2 succeeded at 1590 with fuzz 2 (offset 387 lines). Hunk #3 FAILED at 2374. Hunk #4 succeeded at 2199 with fuzz 1 (offset 181 lines). Hunk #5 succeeded at 2414 (offset 387 lines). Hunk #6 FAILED at 4445. Hunk #7 succeeded at 6278 with fuzz 2 (offset 995 lines). Hunk #8 succeeded at 8692 (offset 1160 lines). Hunk #9 succeeded at 8786 (offset 998 lines). Hunk #10 FAILED at 9624. Hunk #11 succeeded at 9824 with fuzz 2 (offset 1163 lines). Hunk #12 succeeded at 9759 with fuzz 2 (offset 1003 lines). Hunk #13 succeeded at 9928 with fuzz 1 (offset 1163 lines). Hunk #14 succeeded at 9888 with fuzz 2 (offset 1022 lines). Hunk #15 succeeded at 10528 with fuzz 2 (offset 1387 lines). Hunk #16 succeeded at 12007 (offset 1363 lines). Hunk #17 FAILED at 12748. Hunk #18 FAILED at 12764. Hunk #19 FAILED at 12800. Hunk #20 FAILED at 12815. Hunk #21 FAILED at 12840. Hunk #22 succeeded at 12828 (offset 1317 lines). Hunk #23 succeeded at 12931 (offset 1382 lines). Hunk #24 FAILED at 12977. Hunk #25 FAILED at 13040. Hunk #26 succeeded at 12996 (offset 1323 lines). Hunk #27 FAILED at 13070. Hunk #28 succeeded at 16556 (offset 1732 lines). Hunk #29 FAILED at 16580. Hunk #30 succeeded at 18467 (offset 1582 lines). misordered hunks! output would be garbled Hunk #31 FAILED at 18103. Hunk #32 FAILED at 19110. Hunk #33 FAILED at 20636. 16 out of 33 hunks FAILED -- saving rejects to file channels/chan_sip.c.rej patching file channels/chan_skinny.c Hunk #1 FAILED at 2246. Hunk #2 succeeded at 3648 (offset 366 lines). Hunk #3 FAILED at 3805. Hunk #4 FAILED at 3884. Hunk #5 FAILED at 4076. Hunk #6 FAILED at 4116. Hunk #7 FAILED at 4157. Hunk #8 succeeded at 3943 with fuzz 2 (offset 131 lines). 6 out of 8 hunks FAILED -- saving rejects to file channels/chan_skinny.c.rej patching file channels/chan_unistim.c Hunk #1 succeeded at 3672 (offset 6 lines). patching file funcs/func_connectedline.c patching file include/asterisk/channel.h Hunk #1 FAILED at 244. Hunk #2 FAILED at 493. Hunk #3
Re: [asterisk-users] Connected Line ID
Ryan, The problem is not with SIP, but with ISDN. Or is this patch also applied for ISDN calls? Arjan -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Ryan Wagoner Verzonden: 20-06-2011 13:51 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Connected Line ID On Mon, Jun 20, 2011 at 5:39 AM, Arjan Kroon | Mobillion arjan.kr...@mobillion.nl wrote: Oke, But is there a patch from version 1.6.2.12? Greeting, Arjan -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle Verzonden: 20-06-2011 11:36 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Connected Line ID Arjan Kroon | Mobillion wrote: And if I can see it, this patch is already included in version 1.6.2.12. Or am I wrong? That I can't answer. I'm still using 1.4.x and am experimenting with 1.8.x. I recall reading that it wasn't supported directly until 1.8 without patches. Doug I am using 1.8 now, but I had updated the patch for SIPCalledRPID() for 1.6.2 and was using it successfully. http://pastebin.com/K1mmGU1c Ryan -- _ -- 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] Connected Line ID
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner Sent: Monday, June 20, 2011 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Connected Line ID On Mon, Jun 20, 2011 at 5:39 AM, Arjan Kroon | Mobillion arjan.kr...@mobillion.nl wrote: Oke, But is there a patch from version 1.6.2.12? Greeting, Arjan -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle Verzonden: 20-06-2011 11:36 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Connected Line ID Arjan Kroon | Mobillion wrote: And if I can see it, this patch is already included in version 1.6.2.12. Or am I wrong? That I can't answer. I'm still using 1.4.x and am experimenting with 1.8.x. I recall reading that it wasn't supported directly until 1.8 without patches. Doug I am using 1.8 now, but I had updated the patch for SIPCalledRPID() for 1.6.2 and was using it successfully. http://pastebin.com/K1mmGU1c I get the following. What concerns me is the patch unexpectedly ends in middle of line message. [root@rock asterisk-1.6.2.18]# patch --dry-run -p1 -F2 /root/asterisk-1.6.2.10-called-rpid.patch.txt patching file channels/chan_sip.c Hunk #2 succeeded at 1583 (offset 12 lines). Hunk #3 succeeded at 2294 (offset -5 lines). Hunk #4 succeeded at 6762 (offset 210 lines). Hunk #5 succeeded at 10022 (offset 202 lines). Hunk #6 succeeded at 10752 (offset 220 lines). Hunk #7 succeeded at 25914 (offset 345 lines). Hunk #8 succeeded at 25920 (offset 220 lines). Hunk #9 succeeded at 26369 (offset 344 lines). patch unexpectedly ends in middle of line Hunk #10 succeeded at 26309 with fuzz 1 (offset 220 lines). -- _ -- 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] Re : Re : Direct RTP with Asterisk
I didn't think it was possible if the endpoints, or Asterisk was behind a NAT. Someone please correct me if I am wrong. http://www.voip-info.org/wiki/view/Asterisk+sip+directrtpsetup From: Sagbo Romaric Sent: Sun 6/19/2011 9:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Re : Re : Direct RTP with Asterisk I want to build an architecture with client behind NAT without VPN. With can reinvite, the RTP doesn't go directly, I have something like one way audio. Best, De : Roger Burton West ro...@firedrake.org À : asterisk-users@lists.digium.com Envoyé le : Dim 19 juin 2011, 15h 24min 07s Objet : Re: [asterisk-users] Re : Direct RTP with Asterisk On Sun, Jun 19, 2011 at 01:40:31PM +0100, Sagbo Romaric wrote: No, I can't, because, it's a different NAT. I try to simulate P2P with asterisk. What you suggest to me ? I like VPN tunnels. They give you a flat network topology and decent security. -- _ -- 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] Re : Re : Direct RTP with Asterisk
If you can't ping between the two end points, then you can't do direct RTP. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell Sent: Monday, June 20, 2011 8:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re : Re : Direct RTP with Asterisk I didn't think it was possible if the endpoints, or Asterisk was behind a NAT. Someone please correct me if I am wrong. http://www.voip-info.org/wiki/view/Asterisk+sip+directrtpsetup From: Sagbo Romaric Sent: Sun 6/19/2011 9:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Re : Re : Direct RTP with Asterisk I want to build an architecture with client behind NAT without VPN. With can reinvite, the RTP doesn't go directly, I have something like one way audio. Best, De : Roger Burton West ro...@firedrake.org À : asterisk-users@lists.digium.com Envoyé le : Dim 19 juin 2011, 15h 24min 07s Objet : Re: [asterisk-users] Re : Direct RTP with Asterisk On Sun, Jun 19, 2011 at 01:40:31PM +0100, Sagbo Romaric wrote: No, I can't, because, it's a different NAT. I try to simulate P2P with asterisk. What you suggest to me ? I like VPN tunnels. They give you a flat network topology and decent security. -- _ -- 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] Problem with ReceiveFAX app from FFA
On 20/06/2011 8:18 AM, Steve Underwood wrote:
On 06/20/2011 03:38 AM, khalid touati wrote:
Hi Guys,
I solved temporarely my issue by kind of tricking Asterisk, I used
the following line instead of the old:
exten = h,n,System('/usr/local/
bin/fax2mail -p -f ${FAXFILENOEXT} --cid-number ${CALLERID(num)}
--cid-name ${CALLERID(name)} --dest-name Sir/Madam')
now when it hang up I receive my fax through email, and let me tell
you (first time using Free Fax from Asterisk) ReceiveFAX catch well
faxes, just a couple tries but got them all, let's see with more
faxes what will happen.
Why do you consider this a temporary fix? The far end machine will
normally hang up at the end of the FAX, so the hangup option in the
dialplan is exactly where you should expect to be.
I don't know the specifics of how an Asterisk application should exit
however WRT ReceiveFAX() using SPANDSP Technology I would expect the
call to descend to any functions below ReceiveFAX() whether or not the
facsimile was received successfully, the status codes from ReceiveFAX()
can be used by whatever is called next, e.g. a script to e-mail the
received facsimile or a report advising errors were encountered.
I am using a macro to receive faxes, I have placed my System() call back
to were the macro returns after execution due to the function not being
called after ReceiveFAX() under certain conditions.
This however does not guarantee getting an e-mail of what has been
received if the sender decides to abort the transmission.
I can reproduce this using HylaFAX to send a fax to an extension which
Asterisk ReceiveFAX(filename,f) will accept, granted it will fall back
to G.711 mode when receiving, when I abort the transmission using WHFC
client, it is as though ReceiveFAX() goes of somewhere else or simply
decides to forget where it came from as it does not appear to return
hence the System() call is never made.
I should point out I am using extensions.ael for my dialplan.
I personally have considered this behaviour to possibly be a bug.
Cheers,
Larry.
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[asterisk-users] Re : Re : Re : Direct RTP with Asterisk
Dear, Can you provide me the firewall rules which help me to address this issue in the case of my architecture. I try some rules without success. Best, Romaric SAGBO De : Eric Wieling ewiel...@nyigc.com À : Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Envoyé le : Lun 20 juin 2011, 14h 18min 52s Objet : Re: [asterisk-users] Re : Re : Direct RTP with Asterisk If you can't ping between the two end points, then you can't do direct RTP. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell Sent: Monday, June 20, 2011 8:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re : Re : Direct RTP with Asterisk I didn't think it was possible if the endpoints, or Asterisk was behind a NAT. Someone please correct me if I am wrong. http://www.voip-info.org/wiki/view/Asterisk+sip+directrtpsetup From: Sagbo Romaric Sent: Sun 6/19/2011 9:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Re : Re : Direct RTP with Asterisk I want to build an architecture with client behind NAT without VPN. With can reinvite, the RTP doesn't go directly, I have something like one way audio. Best, De : Roger Burton West ro...@firedrake.org À : asterisk-users@lists.digium.com Envoyé le : Dim 19 juin 2011, 15h 24min 07s Objet : Re: [asterisk-users] Re : Direct RTP with Asterisk On Sun, Jun 19, 2011 at 01:40:31PM +0100, Sagbo Romaric wrote: No, I can't, because, it's a different NAT. I try to simulate P2P with asterisk. What you suggest to me ? I like VPN tunnels. They give you a flat network topology and decent security. -- _ -- 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] Re : Re : Direct RTP with Asterisk
On 20/06/11 13:18, Eric Wieling wrote: If you can't ping between the two end points, then you can't do direct RTP. precisely. If 10.10.9.1 isn't reachable from the network that 10.10.8.1 is on then 10.10.8.1 isn't going to be able to send RTP to 10.10.9.1. You need to add routes to the routers on both networks telling them how to reach the other networks. cheers, Paul -- _ -- 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] Integration of OpenVXI
Hi, Have anybody integrated OpenVXI http://www.speech.cs.cmu.edu/openvxi/ with Asterisk? Thanks, Gopal -- _ -- 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] Re : Re : Re : Direct RTP with Asterisk
Ok, thanks, Can you help me to have this kind of rules ? I try with iptables without success. Best, Romaric SAGBO De : Paul Hayes p...@provu.co.uk À : asterisk-users@lists.digium.com Envoyé le : Lun 20 juin 2011, 16h 39min 32s Objet : Re: [asterisk-users] Re : Re : Direct RTP with Asterisk On 20/06/11 13:18, Eric Wieling wrote: If you can't ping between the two end points, then you can't do direct RTP. precisely. If 10.10.9.1 isn't reachable from the network that 10.10.8.1 is on then 10.10.8.1 isn't going to be able to send RTP to 10.10.9.1. You need to add routes to the routers on both networks telling them how to reach the other networks. cheers, Paul -- _ -- 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] Integration of OpenVXI
Check out this product. http://www.i6net.com On Mon, Jun 20, 2011 at 9:40 AM, Gopal krishnan gopalakrishnan...@gmail.com wrote: Hi, Have anybody integrated OpenVXI http://www.speech.cs.cmu.edu/openvxi/ with Asterisk? Thanks, Gopal -- _ -- 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 -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- 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] Re : Re : Re : Direct RTP with Asterisk
The only way this will work is to remove NAT from this scenerio. And it's not Asterisk's fault per se. The phones are built 'that way' also. That's why other free providers don't use SIP phones, but build their own client software. The others are trying to tell you SIP/RTP doesn't work the way you want it to. Lyle Giese LCR Computer Services, Inc. On 06/20/11 10:05, Sagbo Romaric wrote: Ok, thanks, Can you help me to have this kind of rules ? I try with iptables without success. Best, Romaric SAGBO *De :* Paul Hayes p...@provu.co.uk *À :* asterisk-users@lists.digium.com *Envoyé le :* Lun 20 juin 2011, 16h 39min 32s *Objet :* Re: [asterisk-users] Re : Re : Direct RTP with Asterisk On 20/06/11 13:18, Eric Wieling wrote: If you can't ping between the two end points, then you can't do direct RTP. precisely. If 10.10.9.1 isn't reachable from the network that 10.10.8.1 is on then 10.10.8.1 isn't going to be able to send RTP to 10.10.9.1. You need to add routes to the routers on both networks telling them how to reach the other networks. cheers, Paul -- _ -- 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] Re : Re : Re : Direct RTP with Asterisk
You can ask a million more times. The answer will not change. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sagbo Romaric Sent: Monday, June 20, 2011 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Re : Re : Re : Direct RTP with Asterisk Ok, thanks, Can you help me to have this kind of rules ? I try with iptables without success. Best, Romaric SAGBO De : Paul Hayes p...@provu.co.uk À : asterisk-users@lists.digium.com Envoyé le : Lun 20 juin 2011, 16h 39min 32s Objet : Re: [asterisk-users] Re : Re : Direct RTP with Asterisk On 20/06/11 13:18, Eric Wieling wrote: If you can't ping between the two end points, then you can't do direct RTP. precisely. If 10.10.9.1 isn't reachable from the network that 10.10.8.1 is on then 10.10.8.1 isn't going to be able to send RTP to 10.10.9.1. You need to add routes to the routers on both networks telling them how to reach the other networks. cheers, Paul -- _ -- 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] Re : Re : Re : Re : Direct RTP with Asterisk
Now I add route and it's work now. But, I need to improve it because I need to have direct RTP without to have add the rules to firewall. Any client behind his NAT can talk with another behind his NAT. Best for all of you. Romaric SAGBO Ingénieur Réseaux et Télécoms. BP 613 Porto Novo Tél:(+229) 97217745 / 93687458 BENIN De : Lyle Giese l...@lcrcomputer.net À : asterisk-users@lists.digium.com Envoyé le : Lun 20 juin 2011, 17h 19min 05s Objet : Re: [asterisk-users] Re : Re : Re : Direct RTP with Asterisk The only way this will work is to remove NAT from this scenerio. And it's not Asterisk's fault per se. The phones are built 'that way' also. That's why other free providers don't use SIP phones, but build their own client software. The others are trying to tell you SIP/RTP doesn't work the way you want it to. Lyle Giese LCR Computer Services, Inc. On 06/20/11 10:05, Sagbo Romaric wrote: Ok, thanks, Can you help me to have this kind of rules ? I try with iptables without success. Best, Romaric SAGBO *De :* Paul Hayes p...@provu.co.uk *À :* asterisk-users@lists.digium.com *Envoyé le :* Lun 20 juin 2011, 16h 39min 32s *Objet :* Re: [asterisk-users] Re : Re : Direct RTP with Asterisk On 20/06/11 13:18, Eric Wieling wrote: If you can't ping between the two end points, then you can't do direct RTP. precisely. If 10.10.9.1 isn't reachable from the network that 10.10.8.1 is on then 10.10.8.1 isn't going to be able to send RTP to 10.10.9.1. You need to add routes to the routers on both networks telling them how to reach the other networks. cheers, Paul -- _ -- 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 -- _ -- 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] Re : Re : Re : Re : Direct RTP with Asterisk
On 20 Jun 2011, at 16:33, Sagbo Romaric wrote: Any client behind his NAT can talk with another behind his NAT. Still not possible.. The internet doesn't really work like that. SIP even more so. S-- _ -- 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] Integration of OpenVXI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Le 20/06/2011 04:40, Gopal krishnan a écrit : Have anybody integrated OpenVXI http://www.speech.cs.cmu.edu/openvxi/ with Asterisk? Voiceglue works for me: http://www.voiceglue.org/ Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAk3/dbgACgkQuu7Rv+oOo/hemACdEN4qLhxLl9LJGpdGIfd8zZ0B PAsAnRxitrzwt5RhWPeo/iwVuYqfeKNh =LpwD -END PGP SIGNATURE- -- _ -- 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] menu issue
hello liste
i have create an menu like below
exten = my_number,1,Ringing()
exten = my_number,2,Wait(4)
exten = my_number,3,Goto(home,s,1)
[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,2,Background(${sounds_path}welcome)
exten = #,1,Goto(menu,s,1)
exten = i,1,Playback(${sounds_path}error-key)
exten = t,1,Goto(home,s,1)
with this menu i call my_number and i cal listen the welcome message
without issue but when there no key pressed the call hang up
i verify in gool and i found that exten = t,1,Goto(home,s,1) is to send
to the home until i press a key
could you please tell me what is wrong with this menu.
Nb: i use asterisk 1.4
best regards
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Re: [asterisk-users] menu issue
Put this line in
Exten = s,3,goto(home,s,1)
You are experiencing fall through when no dtmf is pressed and since there
is no handling, the call hangs up.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Monday, June 20, 2011 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] menu issue
hello liste
i have create an menu like below
exten = my_number,1,Ringing()
exten = my_number,2,Wait(4)
exten = my_number,3,Goto(home,s,1)
[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,2,Background(${sounds_path}welcome)
exten = #,1,Goto(menu,s,1)
exten = i,1,Playback(${sounds_path}error-key)
exten = t,1,Goto(home,s,1)
with this menu i call my_number and i cal listen the welcome message
without issue but when there no key pressed the call hang up
i verify in gool and i found that exten = t,1,Goto(home,s,1) is to send
to the home until i press a key
could you please tell me what is wrong with this menu.
Nb: i use asterisk 1.4
best regards
--
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Re: [asterisk-users] Polycom BLF
I missed one important parameter in my setup of BLF for polycom phones (at
least if you want to do one touch directed pickup)
In sip.conf add
notifycid=yes
the notifycid=yes causes asterisk to add a target uri = callID to the XML
of the SIP notify. Without this target uri the Polycom phone will not do a
directed pickup.
On Fri, Jun 17, 2011 at 2:17 PM, Gord Urquhart gord...@gmail.com wrote:
From http://www.voip-info.org/wiki/view/Asterisk+presence
Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With
SIP 3.2.X firmware (available on the Polycom download site) and Asterisk
1.6.1, Polycom phones now support a full featured BLF showing statuses of
Ringing, Inuse and Online and one touch directed call pickup.
On the asterisk side all that needs to be done is to add a hint to the
extension and enable directed pickup. Directed pickup is enabled by adding
the following lines to extensios.conf
exten = _*8.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten = _*8.,n,Pickup(${EXTEN:2}@PICKUPMARK)
On the phone side for each line that is going to be monitored add lines
like the following to the phone's cfg file.
attendant.reg=1
attendant.resourceList.1.address=sip:205@192.168.1.102
attendant.resourceList.1.label=205
attendant.resourceList.2.address=sip:217@192.168.1.102
attendant.resourceList.2.label=217
call.directedCallPickupMethod=legacy
call.directedCallPickupString=*8
feature.12.name=directed-call-pickup
feature.12.enabled=1
Assuming my server is at 192.168.1.102, this will add two BLF lines to the
phone for extensions 205 and 217. Calls incoming to those extensions will
show a blinking green led on the monitoring phone, pressing the hard key
will pick the call up, if it is answered elsewhere the led will change to
solid red. AFAIK this cannot be configured via the phones web gui, you must
use the cfg files. You can also use versions of Asterisk older than 1.6.1 if
you remove the restriction on what asterisk thinks Polycom phones can
handle. Look in chan_sip.c for
if (strstr(p-useragent, Polycom)) {
p-subscribed = XPIDF_XML;
and change that line to
p-subscribed = DIALOG_INFO_XML;
On Tue, Jun 14, 2011 at 8:36 AM, Jeff LaCoursiere j...@sunfone.comwrote:
Struggling with an IP650 and 7 IP335s this morning. I have the following
hints defined (courtesy of FreePBX 2.9):
extensions_additional.conf:**exten = 300,hint,SIP/300
extensions_additional.conf:**exten = 301,hint,SIP/301
extensions_additional.conf:**exten = 302,hint,SIP/302
extensions_additional.conf:**exten = 303,hint,SIP/303
extensions_additional.conf:**exten = 304,hint,SIP/304
extensions_additional.conf:**exten = 305,hint,SIP/305
extensions_additional.conf:**exten = 307,hint,SIP/307
extensions_additional.conf:**exten = 308,hint,SIP/308
extensions_additional.conf:**exten = 322,hint,SIP/322
extensions_additional.conf:**exten = 350,hint,SIP/350
extensions_additional.conf:**exten = 400,hint,SIP/400
The Polycoms are all pulling an XML directory via FTP where each extension
has BW (Buddy Watch) set to 1:
item
lnMehra/ln
fnRay/fn
ct301/ct
sd101/sd
bw1/bw
/item
This all actually works fine, and from the reception phone (the 650) I can
see the status of all the extensions, and if I dig into some menus on the
335 I can see status as well. So I would expect that core show hints
would show '8' for all extensions, but it doesn't:
artha*CLI core show hints
-= Registered Asterisk Dial Plan Hints =-
300@ext-local : SIP/300 State:Idle
Watchers 7
301@ext-local : SIP/301 State:Idle
Watchers 8
302@ext-local : SIP/302 State:Idle
Watchers 8
303@ext-local : SIP/303 State:Idle
Watchers 8
304@ext-local : SIP/304 State:InUse
Watchers 8
305@ext-local : SIP/305 State:Idle
Watchers 7
307@ext-local : SIP/307 State:Idle
Watchers 1
308@ext-local : SIP/308 State:Idle
Watchers 7
350@ext-local : SIP/350 State:Idle
Watchers 1
400@ext-local : SIP/400 State:InUse
Watchers 7
- 11 hints registered
Something seems broken here. And the 650 seems to lose its hint for a
phone once in a while, and report it as unreachable, even though it can
easily make and receive calls from it.
Am I tilting at windmills? Is this really unstable or has someone made it
work solidly?
Thanks!
--
Jeff LaCoursiere
SunFone
[asterisk-users] Problems with pickupgroup/callgroup with Asterisk 1.8.4.2
I have problems using the call pickup under Asterisk 1.8.4.2. I have another Asterisk with 1.6 - and it is working fine with the same settings. I have setup the same callgroup and pickupgroup for all extensions in sip.conf - just to make things simple for testing. The sequence *8 seems to be completely ignored by Asterisk - the client shows Call answered when dialing *8 while the other phones keep on ringing (both software and hard phone clients) - but I see nothing in the Asterisk console. It's like the *8 sequence skips Asterisk and goes through the iax trunks straight upstream to the trunks provider. Then weird messages show up in the console about Max retries exceeded to host, I can't use our IAX outgoing trunks and the only way to get things working again is to restart Asterisk. Am I missing something silly here? Here is my sip.conf: [general] subscribecontext=sip-blf context=default disallow=all allow=alaw allow=ulaw allowguest=no tcpenable=no tlsenable=no srvlookup=no localnet=192.168.56.0/255.255.255.0 localnet=192.168.57.0/255.255.255.0 tos_sip=cs3; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. canreinvite=no dtmfmode = rfc2833 notifyringing=yes limitonpeers=yes callcounter=yes [basic-phone](!) type=friend context=from_internal_phones nat=no qualify=yes host=dynamic mohinterpret=default mohsuggest=default call-limit=20 callgroup=1 pickupgroup=1 [21](basic-phone) secret=mypassword [22](basic-phone) secret=mypassword [200](basic-phone) secret=mypassword And here is a trace of a call coming in through the IAX trunk, ringing internal sip phones 21 and 22, while I try to pick it up from 200: Connected to Asterisk 1.8.4.2 currently running on khca-server (pid = 3915) Verbosity was 3 and is now 6 -- Accepting AUTHENTICATED call from 111.222.333.444 : requested format = alaw, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw), priority = mine -- Executing [0123456@from_trunks:1] SIPAddHeader(IAX2/khca_in-2443, Alert-I nfo: http://127.0.0.1\;info=external) in new stack -- Executing [0123456@from_trunks:2] Goto(IAX2/khca_in-2443, -- Executing [0123456@from_trunks:3] Dial(IAX2/khca_in-2443, SIP/21SIP/22SIP/2 3,,t) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called 21 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called 22 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called 23 -- SIP/21- is ringing -- SIP/23-0002 is ringing -- SIP/22-0001 is ringing == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 [Jun 20 18:41:56] WARNING[3936]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 11, ts=20015, seqno=5) [Jun 20 18:41:57] WARNING[3934]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 2, ts=21015, seqno=6) [Jun 20 18:42:06] WARNING[3938]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 11, ts=30014, seqno=7) [Jun 20 18:42:16] WARNING[3938]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 11, ts=40014, seqno=8) [Jun 20 18:42:18] WARNING[3939]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 2, ts=42014, seqno=9) [Jun 20 18:42:26] WARNING[3931]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 11, ts=50014, seqno=10) [Jun 20 18:42:36] WARNING[3932]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 11, ts=60013, seqno=11) [Jun 20 18:42:39] WARNING[3931]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 2, ts=63014, seqno=12) [Jun 20 18:42:46] WARNING[3939]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 11, ts=70013, seqno=13) -- _ -- 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] Asterisk call limitation
Dears, i am using sipp to test asterisk(1.6.22) performance ,but when i limit the calls to 150 ,only 100 active calls on asterisk found ?why sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: mailto:kche...@xplorium.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com attachment: winmail.dat-- _ -- 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 call limitation
On 06/20/2011 01:09 PM, Khaled W. Chehab wrote: Dears, i am using sipp to test asterisk(1.6.22) performance ,but when i limit the calls to 150 ,only 100 active calls on asterisk found ?why sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 You did not provide any log output, or anything that could be used to try to help you understand your problem. Without any details, any reply you get would be just a guess, nothing more. Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail:mailto:kche...@xplorium.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com Please refrain from including 20-line signature blocks in your messages to the Asterisk mailing lists (or really, anywhere). Your message had three lines of content and 30+ lines of non-content. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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] Cisco IP Phones and Skinny in asterisk
On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; snip Have you thought about perhaps just flashing the phones to use the SIP firmware? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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] SMS with Asterisk
On Mon, Jun 20, 2011 at 3:52 AM, Steve Totaro stot...@asteriskhelpdesk.comwrote: Two requests, not from me but the community. 1. Don't top post *cough* 2. When you find your solution, reply to this thread so others will be (silver) spoon fed the answers and blindly accept them without trying things and going through a learning curve and experimentation when they find your post in Google. I hear some people are actually deploying their asterisk solutions in war zones and are taking heavy fire while they're looking for answers - seems like it would make their life a whole lot easier (and safer!) if people posted simple responses on this list when suggestions worked for them... -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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] Cisco IP Phones and Skinny in asterisk
I'm using the sip firmware.. It's alright.. I feel like I'm not receiving all the features I should.. But MWI works and multiple call appearance.. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Monday, June 20, 2011 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; snip Have you thought about perhaps just flashing the phones to use the SIP firmware? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.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] Problem with ReceiveFAX app from FFA
On Mon, Jun 20, 2011 at 7:44 AM, Larry Moore lmo...@starwon.com.au wrote: snip I personally have considered this behaviour to possibly be a bug. Once a fax is sent, the sending fax machine typically hangs up the call - sending the call to the h extension. It's the same as if you are on an actual call that was connected using the Dial() application, and the other end hangs up - the next step is the 'h' extension, not to continue in the current dialplan. I don't see how this is a bug, unless you think the entire call-flow paradigm that currently exists in asterisk is a bug. Now, if you're not getting certain variables to pass into the 'h' extension, that you feel should indeed be passed into the 'h' extension, that may be considered a bug...but you would need to show us CLI output and existing dialplan for followup. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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] menu issue
On Mon, Jun 20, 2011 at 12:17 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,2,Background(${sounds_path}welcome)
exten = #,1,Goto(menu,s,1)
exten = i,1,Playback(${sounds_path}error-key)
exten = t,1,Goto(home,s,1)
You need to add the following to the [home] context:
exten = s,3,WaitExten(10)
which will cause the call to wait 10 seconds for input, otherwise it will
timeout and go to the 't' extension. The way you currently have it, the
call will end after the Background() app finishes playing because it has no
additional steps and nothing that will tell it to go to the 't' extension.
Also, consider switching your dialplan priorities away from 1,2,3... and
go to 1,n,n,n... as this reduces headaches in the longrun.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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[asterisk-users] Get second cipher in an extension
Hello list, how can I get the second character/cipher of an extension ? If I have : exten = 12345,n,NoOP() How can I get 2 ? If I have : exten = 787,n,NoOP() How can I get 8 ? Thanks ! Kind regards, 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] Get second cipher in an extension
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Monday, June 20, 2011 3:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Get second cipher in an extension Hello list, how can I get the second character/cipher of an extension ? If I have : exten = 12345,n,NoOP() How can I get 2 ? If I have : exten = 787,n,NoOP() How can I get 8 ? https://wiki.asterisk.org/wiki/display/AST/Selecting+Characters+from+Variables -- _ -- 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] Get second cipher in an extension
how can I get the second character/cipher of an extension ?
If I have : exten = 12345,n,NoOP()
How can I get 2 ?
${EXTEN:1:1}
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Re: [asterisk-users] Get second cipher in an extension
On Mon, Jun 20, 2011 at 2:09 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
**
Hello list,
how can I get the second character/cipher of an extension ?
snip
I vaguely recall that to get a substring out of an extension variable, you
would use it in the format ${EXTEN:offset:length}, so for your example it
would be
${EXTEN:1:1}
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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Re: [asterisk-users] Problems with pickupgroup/callgroup with Asterisk 1.8.4.2
Replying to my own post: I have done some more digging, disabling parts of configuration files one at a time - since there is nothing useful in the console for this problem. Turns out that if I enable the following lines in features.conf: parkext = 700 parkpos = 701-720 context = parkedcalls then I can't pick calls up with *8 anymore. Here is everything I have enabled in the features.conf, just in case: [general] parkext = 700 parkpos = 701-720 context = parkedcalls pickupexten = *8 [featuremap] disconnect = *0 Can anybody think of any reason for this? Could it be something I'm doing - or should I report it as a bug? Sebastian On 20/06/11 19:00, Sebastian Arcus wrote: I have problems using the call pickup under Asterisk 1.8.4.2. I have another Asterisk with 1.6 - and it is working fine with the same settings. I have setup the same callgroup and pickupgroup for all extensions in sip.conf - just to make things simple for testing. The sequence *8 seems to be completely ignored by Asterisk - the client shows Call answered when dialing *8 while the other phones keep on ringing (both software and hard phone clients) - but I see nothing in the Asterisk console. It's like the *8 sequence skips Asterisk and goes through the iax trunks straight upstream to the trunks provider. Then weird messages show up in the console about Max retries exceeded to host, I can't use our IAX outgoing trunks and the only way to get things working again is to restart Asterisk. Am I missing something silly here? Here is my sip.conf: [general] subscribecontext=sip-blf context=default disallow=all allow=alaw allow=ulaw allowguest=no tcpenable=no tlsenable=no srvlookup=no localnet=192.168.56.0/255.255.255.0 localnet=192.168.57.0/255.255.255.0 tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. canreinvite=no dtmfmode = rfc2833 notifyringing=yes limitonpeers=yes callcounter=yes [basic-phone](!) type=friend context=from_internal_phones nat=no qualify=yes host=dynamic mohinterpret=default mohsuggest=default call-limit=20 callgroup=1 pickupgroup=1 [21](basic-phone) secret=mypassword [22](basic-phone) secret=mypassword [200](basic-phone) secret=mypassword And here is a trace of a call coming in through the IAX trunk, ringing internal sip phones 21 and 22, while I try to pick it up from 200: Connected to Asterisk 1.8.4.2 currently running on khca-server (pid = 3915) Verbosity was 3 and is now 6 -- Accepting AUTHENTICATED call from 111.222.333.444 : requested format = alaw, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw), priority = mine -- Executing [0123456@from_trunks:1] SIPAddHeader(IAX2/khca_in-2443, Alert-I nfo: http://127.0.0.1\;info=external) in new stack -- Executing [0123456@from_trunks:2] Goto(IAX2/khca_in-2443, -- Executing [0123456@from_trunks:3] Dial(IAX2/khca_in-2443, SIP/21SIP/22SIP/2 3,,t) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called 21 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called 22 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called 23 -- SIP/21- is ringing -- SIP/23-0002 is ringing -- SIP/22-0001 is ringing == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 [Jun 20 18:41:56] WARNING[3936]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 11, ts=20015, seqno=5) [Jun 20 18:41:57] WARNING[3934]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 2, ts=21015, seqno=6) [Jun 20 18:42:06] WARNING[3938]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 11, ts=30014, seqno=7) [Jun 20 18:42:16] WARNING[3938]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 11, ts=40014, seqno=8) [Jun 20 18:42:18] WARNING[3939]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 2, ts=42014, seqno=9) [Jun 20 18:42:26] WARNING[3931]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 11, ts=50014, seqno=10) [Jun 20 18:42:36] WARNING[3932]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 11, ts=60013, seqno=11) [Jun 20 18:42:39] WARNING[3931]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 2, ts=63014, seqno=12) [Jun 20 18:42:46] WARNING[3939]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 212.11.91.204 on IAX2/khca_in-2443 (type = 6, subclass = 11, ts=70013, seqno=13) -- _ -- Bandwidth
Re: [asterisk-users] Problem with ReceiveFAX app from FFA
From: Warren Selby wcse...@selbytech.com Sent: Monday, June 20, 2011 3:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problem with ReceiveFAX app from FFA On Mon, Jun 20, 2011 at 7:44 AM, Larry Moore lmo...@starwon.com.au wrote: snip I personally have considered this behaviour to possibly be a bug. Once a fax is sent, the sending fax machine typically hangs up the call - sending the call to the h extension. It's the same as if you are on an actual call that was connected using the Dial() application, and the other end hangs up - the next step is the 'h' extension, not to continue in the current dialplan. I don't see how this is a bug, unless you think the entire call-flow paradigm that currently exists in asterisk is a bug. Now, if you're not getting certain variables to pass into the 'h' extension, that you feel should indeed be passed into the 'h' extension, that may be considered a bug...but you would need to show us CLI output and existing dialplan for followup. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com Waren I concur we use the h extension to log inbound faxes to a database and then we process them outside the asterisk platform. Our biggest issue with ReceiveFAX is about a 20% t.38 negotiation fail ratio. We then force fall back to t.30 for the next call from that number. We would like to see better success with t.38. Today our primary server has had 910 faxes of which 707 negotiated t.38, 44 have failed darn robo dialers, The rest failed the first attempt and came in T.30 on the second call. 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] Asterisk call limitation
It could be your OS limit try ulimit command. -- Sent from my iPhone On Jun 20, 2011, at 2:21 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 06/20/2011 01:09 PM, Khaled W. Chehab wrote: Dears, i am using sipp to test asterisk(1.6.22) performance ,but when i limit the calls to 150 ,only 100 active calls on asterisk found ?why sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 You did not provide any log output, or anything that could be used to try to help you understand your problem. Without any details, any reply you get would be just a guess, nothing more. Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail:mailto:kche...@xplorium.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com Please refrain from including 20-line signature blocks in your messages to the Asterisk mailing lists (or really, anywhere). Your message had three lines of content and 30+ lines of non-content. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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
Re: [asterisk-users] Problem with ReceiveFAX app from FFA
On Mon, Jun 20, 2011 at 2:43 PM, Bryant Zimmerman brya...@zktech.comwrote: I concur we use the h extension to log inbound faxes to a database and then we process them outside the asterisk platform. Our biggest issue with ReceiveFAX is about a 20% t.38 negotiation fail ratio. We then force fall back to t.30 for the next call from that number. We would like to see better success with t.38. Today our primary server has had 910 faxes of which 707 negotiated t.38, 44 have failed darn robo dialers, The rest failed the first attempt and came in T.30 on the second call I'm not sure how much of this is the fault of FFA versus the fault of shoddy t.38 implementations out in the wild. I've had a ton of headaches trying to get t.38 solutions implemented with various ITSP's and FFA. I've heard that the free SpanDSP version has better negotiation rates, however, I have not personally tested them. In the end, for mission critical fax applications (yes, these still exist, especially in the financial sector), I tend to go with a dedicated line and DID used in conjunction with an FXO device or T1 device, an IAXModem connection over a local, low-latency LAN, and setup a dialplan pass-through to a hylafax server. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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] Asterisk call limitation
I tried the ulimit [root@localhost ~]# ulimit Unlimited Then sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) 100 active channels 100 active calls 6407 calls processed [root@localhost ~]# I find in /var/log/asterisk/full [Jun 20 09:43:17] NOTICE[9705] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Jun 20 09:43:17] VERBOSE[3068] chan_unistim.c: Reloading unistim.conf... [Jun 20 16:43:33] WARNING[12353] file.c: Failed to write frame [Jun 20 16:43:34] WARNING[12389] file.c: Failed to write frame [Jun 20 16:43:35] WARNING[12394] file.c: Failed to write frame [Jun 20 16:43:43] WARNING[12484] file.c: Failed to write frame [Jun 20 16:43:44] WARNING[12488] file.c: Failed to write frame [Jun 20 16:43:52] WARNING[12573] file.c: Failed to write frame [Jun 20 16:43:57] WARNING[12625] file.c: Failed to write frame [Jun 20 16:44:07] WARNING[12723] file.c: Failed to write frame [Jun 20 16:44:14] WARNING[12789] file.c: Failed to write frame [Jun 20 16:44:22] WARNING[12872] file.c: Failed to write frame [Jun 20 16:44:26] WARNING[12908] file.c: Failed to write frame Khaled Chehab NGN Eng. Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Monday, June 20, 2011 11:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk call limitation It could be your OS limit try ulimit command. -- Sent from my iPhone On Jun 20, 2011, at 2:21 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 06/20/2011 01:09 PM, Khaled W. Chehab wrote: Dears, i am using sipp to test asterisk(1.6.22) performance ,but when i limit the calls to 150 ,only 100 active calls on asterisk found ?why sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 You did not provide any log output, or anything that could be used to try to help you understand your problem. Without any details, any reply you get would be just a guess, nothing more. Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail:mailto:kche...@xplorium.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com Please refrain from including 20-line signature blocks in your messages to
Re: [asterisk-users] Asterisk call limitation
Oh! Wait you set ulimit for running shellYou should set ulimit on asterisk. Also you can set ulimit command on asterisk startup file / etc/init.d/asterisk and restart asterisk also you can set in limit.conf file I had this issue before and I solved that way. -- Sent from my iPhone On Jun 20, 2011, at 4:47 PM, Khaled W. Chehab kche...@xplorium.com wrote: I tried the ulimit [root@localhost ~]# ulimit Unlimited Then sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) 100 active channels 100 active calls 6407 calls processed [root@localhost ~]# I find in /var/log/asterisk/full [Jun 20 09:43:17] NOTICE[9705] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Jun 20 09:43:17] VERBOSE[3068] chan_unistim.c: Reloading unistim.conf... [Jun 20 16:43:33] WARNING[12353] file.c: Failed to write frame [Jun 20 16:43:34] WARNING[12389] file.c: Failed to write frame [Jun 20 16:43:35] WARNING[12394] file.c: Failed to write frame [Jun 20 16:43:43] WARNING[12484] file.c: Failed to write frame [Jun 20 16:43:44] WARNING[12488] file.c: Failed to write frame [Jun 20 16:43:52] WARNING[12573] file.c: Failed to write frame [Jun 20 16:43:57] WARNING[12625] file.c: Failed to write frame [Jun 20 16:44:07] WARNING[12723] file.c: Failed to write frame [Jun 20 16:44:14] WARNING[12789] file.c: Failed to write frame [Jun 20 16:44:22] WARNING[12872] file.c: Failed to write frame [Jun 20 16:44:26] WARNING[12908] file.c: Failed to write frame Khaled Chehab NGN Eng. Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Monday, June 20, 2011 11:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk call limitation It could be your OS limit try ulimit command. -- Sent from my iPhone On Jun 20, 2011, at 2:21 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 06/20/2011 01:09 PM, Khaled W. Chehab wrote: Dears, i am using sipp to test asterisk(1.6.22) performance ,but when i limit the calls to 150 ,only 100 active calls on asterisk found ?why sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 You did not provide any log output, or anything that could be used to try to help you understand your problem. Without any details, any reply you get would be just a guess, nothing more. Regards
Re: [asterisk-users] Asterisk call limitation
Can you please specify more 1-how to set the ulimit on [root@localhost ~]# ulimit unlimited [root@localhost ~]# ulimit --help -bash: ulimit: --: invalid option ulimit: usage: ulimit [-SHacdfilmnpqstuvx] [limit] - How to set the ulimit command on in /etc/init.d/asterisk Since there is no parameter for ulimit in the file Thanks in advance Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Tuesday, June 21, 2011 12:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk call limitation Oh! Wait you set ulimit for running shellYou should set ulimit on asterisk. Also you can set ulimit command on asterisk startup file / etc/init.d/asterisk and restart asterisk also you can set in limit.conf file I had this issue before and I solved that way. -- Sent from my iPhone On Jun 20, 2011, at 4:47 PM, Khaled W. Chehab kche...@xplorium.com wrote: I tried the ulimit [root@localhost ~]# ulimit Unlimited Then sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) 100 active channels 100 active calls 6407 calls processed [root@localhost ~]# I find in /var/log/asterisk/full [Jun 20 09:43:17] NOTICE[9705] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Jun 20 09:43:17] VERBOSE[3068] chan_unistim.c: Reloading unistim.conf... [Jun 20 16:43:33] WARNING[12353] file.c: Failed to write frame [Jun 20 16:43:34] WARNING[12389] file.c: Failed to write frame [Jun 20 16:43:35] WARNING[12394] file.c: Failed to write frame [Jun 20 16:43:43] WARNING[12484] file.c: Failed to write frame [Jun 20 16:43:44] WARNING[12488] file.c: Failed to write frame [Jun 20 16:43:52] WARNING[12573] file.c: Failed to write frame [Jun 20 16:43:57] WARNING[12625] file.c: Failed to write frame [Jun 20 16:44:07] WARNING[12723] file.c: Failed to write frame [Jun 20 16:44:14] WARNING[12789] file.c: Failed to write frame [Jun 20 16:44:22] WARNING[12872] file.c: Failed to write frame [Jun 20 16:44:26] WARNING[12908] file.c: Failed to write frame Khaled Chehab NGN Eng. Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com -Original Message- From: asterisk-users-boun...@lists.digium.com
Re: [asterisk-users] Queue Log in Mysql
Sorry, to not answer before! Thanks a lot, as sun as i am able i will test this setup! []'sf.rique On Fri, Jun 17, 2011 at 4:50 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On Thu, 2011-06-16 at 19:12 -0300, Henrique Fernandes wrote: It is possible to log queue in mysql without turning on realtime asterisk? Thanks! []'sf.rique -- Hi Yes, you can pick and choose which things you want to use your DB by defining them in your extconfig.conf so, in extconfig.conf you would need to add queue_log=mysql,your-db-name,queue_log in res_config_mysql.conf (1.8) or res_mysql.conf (1.4,1.6) you would have to put in the connection details for your database If you are using 1.8 your table create statement would be CREATE TABLE `queue_log` ( `id` int(10) unsigned NOT NULL auto_increment, `time` char(26) default NULL, `callid` varchar(32) NOT NULL default '', `queuename` varchar(32) NOT NULL default '', `agent` varchar(32) NOT NULL default '', `event` varchar(32) NOT NULL default '', `data` varchar(100) NOT NULL default '', `data1` VARCHAR(100), `data2` VARCHAR(100), `data3` VARCHAR(100), `data4` VARCHAR(100), `data5` VARCHAR(100), PRIMARY KEY (`id`) )ENGINE=InnoDB ; Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 call limitation
The problem remains even when I add to /etc/init.d/asterisk ulimit -n 65536 [root@localhost ~]# ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 65536 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 1024 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 65536 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited [root@localhost ~]# -Original Message- From: Khaled W. Chehab [mailto:kche...@xplorium.com] Sent: Tuesday, June 21, 2011 12:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk call limitation Can you please specify more 1-how to set the ulimit on [root@localhost ~]# ulimit unlimited [root@localhost ~]# ulimit --help -bash: ulimit: --: invalid option ulimit: usage: ulimit [-SHacdfilmnpqstuvx] [limit] - How to set the ulimit command on in /etc/init.d/asterisk Since there is no parameter for ulimit in the file Thanks in advance Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Tuesday, June 21, 2011 12:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk call limitation Oh! Wait you set ulimit for running shellYou should set ulimit on asterisk. Also you can set ulimit command on asterisk startup file / etc/init.d/asterisk and restart asterisk also you can set in limit.conf file I had this issue before and I solved that way. -- Sent from my iPhone On Jun 20, 2011, at 4:47 PM, Khaled W. Chehab kche...@xplorium.com wrote: I tried the ulimit [root@localhost ~]# ulimit Unlimited Then sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) 100 active channels 100 active calls 6407 calls processed [root@localhost ~]# I find in /var/log/asterisk/full [Jun 20 09:43:17] NOTICE[9705] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Jun 20 09:43:17] VERBOSE[3068] chan_unistim.c: Reloading unistim.conf... [Jun 20
Re: [asterisk-users] No audio after a reinvite changing codec
Inviato da iPhone Il giorno 18/giu/2011, alle ore 06:40, Larry Moore lmo...@starwon.com.au ha scritto: On 18/06/2011 5:36 AM, Matteo Campana wrote: Inviato da iPhone Il giorno 16/giu/2011, alle ore 16:37, Eric Wielingewiel...@nyigc.com ha scritto: We experience the same thing. The solution we use is to not change codecs in the middle of a call. I assumed it was an issue with our upstream. Hi Eric, this behavior is an asterisk bug or asterisk can never change the codec on the fly? Thanks, Matteo The problem reported occurs after a fax tone is detected, one might expect T.38 or G711 to be used to handle the fax, not G729! There is no configuration file information for each of the nodes/peers, no debug of each peer involved nor a trace of the packets hence no one will have ideas! Larry. Hi, I'm out of the office this week, next Monday I will send the debug to the list. However I think It's strange asterisk behavior: it says 200 OK after a re-invite by the provider, but stops to send rtp. Regards, Matteo -- _ -- 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] Cisco IP Phones and Skinny in asterisk
Dear Stefan; First of all, I tried skinny and I faced two major problems (so if I am going to face same problems in sccp then no need to use sccp, so please advise). The two problems that I faced them are: 1) When I do reload then the skinny channel is reloaded and that will cause a restart for the Cisco IP Phones (that are registered to skinny channel). Is the same thing happening with u when u r using sccp channel? 2) When I called the Phone, it is ringing, when we pickup the handset to answer the call, we hear t and we do not hear what source is talking and source does not hear us even .. but if we select music on hold, then caller will hear the music. Also, when we tried to use the Ciscp IP Phone to place a call, while we are dialing, the too tone is always existed and it is ringing at destination but no voice (always t). So, with sccp no problem? From the other side, if I need to use sccp (if we assumed the above problems are not existed) then can u please help for below: 1) If i used sccp and I gave the IP Phone the IP address TFTP server, and no configuration files were existed on TFTP, then it will register on the asterisk sccp channel? 2) The sccp.conf file, where I can find it? Is it the same as the skinny.conf file? 3) To use sccp instead of the skinny channel, all what I need is to unload the skinny from the modules.conf file and load the sccp channel in the modules.conf, and I can use the skinny.conf file for the configuration? About the firmware on the Phone, it will stay the same? I appreciate the kindly help please. Regards Bilal --- Hi, On 06/13/2011 01:04 PM, bilal ghayyad wrote: Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? SCCP works better than SIP in my opinion as there are more features. Check out http://chan-sccp-b.sourceforge.net/ -- _ -- 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] Cisco IP Phones and Skinny in asterisk
If I need to use SIP, from where to get the suitable firmware for these Cisco IP Phones 7942G? Where do u download the SIP firmware usually for your Cisco IP Phones? Your kindly help is highly appreciated. Regards Bilal --- I'm using the sip firmware.. It's alright.. I feel like I'm not receiving all the features I should.. But MWI works and multiple call appearance.. On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; snip Have you thought about perhaps just flashing the phones to use the SIP firmware? -- 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
[asterisk-users] pickupsound = beep kills call pickup in Asterisk 1.8.4.2
I have discovered that if I enable pickupsound = beep in features.conf, if I try to do a pickup with *8, the calling channel keeps on ringing, while the phone where I pick-up from shows that the call has been answered (I don't know where though). Also, it seems to completely bugger up my outgoing IAX trunk (I really can't see the connection, as I'm doing pick-up for a SIP channel). I can only shut Asterisk down with killall asterisk -s9 - nothing else works. I've tried starting the console with asterisk -rvv - but there is nothing unusual there. Could someone please confirm this behaviour on their box, before I go and submit a bug - in case I am doing something wrong? As soon as I comment out pickupsound = beep - everything works just fine and I can do call pickup with *8. Sebastian -- _ -- 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] Cisco IP Phones and Skinny in asterisk
You are supposed to go via cisco and support contract BUT Google is your friend (JFGI) Sent from my iPhone On Jun 20, 2011, at 6:44 PM, bilal ghayyad bilmar...@yahoo.com wrote: If I need to use SIP, from where to get the suitable firmware for these Cisco IP Phones 7942G? Where do u download the SIP firmware usually for your Cisco IP Phones? Your kindly help is highly appreciated. Regards Bilal --- I'm using the sip firmware.. It's alright.. I feel like I'm not receiving all the features I should.. But MWI works and multiple call appearance.. On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; snip Have you thought about perhaps just flashing the phones to use the SIP firmware? -- 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] pickupsound = beep kills call pickup in Asterisk1.8.4.2
This has been fixed only last month, see https://issues.asterisk.org/view.php?id=18654 and try bug18654.diff.txt That will avoid the deadlock, but it's not the proper fix, there are other issues that could trip you up, mainly to do with race conditions with multiple channels picking up the same ringing extensions. Alec Davis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Arcus Sent: Tuesday, 21 June 2011 11:10 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] pickupsound = beep kills call pickup in Asterisk1.8.4.2 I have discovered that if I enable pickupsound = beep in features.conf, if I try to do a pickup with *8, the calling channel keeps on ringing, while the phone where I pick-up from shows that the call has been answered (I don't know where though). Also, it seems to completely bugger up my outgoing IAX trunk (I really can't see the connection, as I'm doing pick-up for a SIP channel). I can only shut Asterisk down with killall asterisk -s9 - nothing else works. I've tried starting the console with asterisk -rvv - but there is nothing unusual there. Could someone please confirm this behaviour on their box, before I go and submit a bug - in case I am doing something wrong? As soon as I comment out pickupsound = beep - everything works just fine and I can do call pickup with *8. Sebastian -- _ -- 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] Inbound CallerID displays asterisk
I have an asterisk 1.4.26 mte running. Sometimes inbound caller ID displays asterisk These calls do not show up on the CLI nor the CDR. I read somewhere that these are asterisk hack attempts. Is this true? What is the best way to defend from this? I know a secure password and all but the client is getting annoyed with the random inbound calls. Thanks, --E -- _ -- 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] Cisco IP Phones and Skinny in asterisk
On Mon, Jun 20, 2011 at 6:10 PM, Robert-iPhone rhuddles...@gmail.comwrote: You are supposed to go via cisco and support contract BUT Google is your friend (JFGI) The support contract from Cisco is only US $8.99 on CDW I really hate to link to my own blog, but I do have a post on there that details how to setup a 79x1 phone using SIP firmware with asterisk. Click the link in my signature and go to the Blog and you should be able to easily find the relevant post. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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] Inbound CallerID displays asterisk
On Mon, Jun 20, 2011 at 6:33 PM, ERIC HERRON e...@lanline.com wrote: I have an asterisk 1.4.26 mte running. ** ** Sometimes inbound caller ID displays “asterisk” ** ** These calls do not show up on the CLI nor the CDR. ** ** I read somewhere that these are asterisk hack attempts. ** ** Is this true? ** ** What is the best way to defend from this? ** ** I know a secure password and all but the client is getting annoyed with the random inbound calls. ** ** Crank you CLI verbosity up to 10 or so and wait for the next time this happens. You should see SOMETHING on the CLI during the call. Post that output to the list and we can help you from there. This does not always indicate someone attempting to hack you, I've seen this occur when there are line errors on FXO devices (among other things). -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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] : Re: ITSP failover for PRI
On Mon, Jun 20, 2011 at 8:38 PM, Claude Hayn chayn...@gmail.com wrote: snip Can someone please make suggestions or point us in the right direction to resolve this no audio issue? No audio is usually a NAT issue. Verify you have the proper NAT settings on your ITSP2 account settings and try again. A SIP debug trace would be very useful for debugging this (sip set debug on on the asterisk CLI or tcpdump -l -n -s 0 -w sipdebug.pcap port 5060 from the command line). -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko
Sent from my iPhone -- _ -- 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] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko
I nominate this for most imaginative use of Asterisk-users of 2011. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jun 20, 2011, at 8:43 PM, Marcelo marcelol...@gmail.com wrote: Sent from my iPhone -- _ -- 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] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko
On Mon, Jun 20, 2011 at 11:47 PM, Alex Balashov abalas...@evaristesys.com wrote: I nominate this for most imaginative use of Asterisk-users of 2011. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jun 20, 2011, at 8:43 PM, Marcelo marcelol...@gmail.com wrote: Sent from my iPhone -- butt dial FTW -- ~~~ Andrew lathama Latham lath...@gmail.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] Looking for Sipura-2000 Latest Firmware
Dear Asterisk Users, I have a Sipura 2000 device, and since last few days I have been searching for its latest firmware for upgrade. Googling tells me that Cisco has stopped the support for this device and I dont have definite idea on where would I be able to find the firmware to upgrade my device. Any help in regards to getting the firmware will be helpful. Regards, Amol -- _ -- 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 Sipura-2000 Latest Firmware
Dear all, New day has brought me luck :) I got the solution. Please find the link for the upgrades. I will try it at my end and if it doesnt work will inform the thread otherwise will not disturb you. http://www.quickconnectusa.com/resources/sipura.asp Cheers, Amol On Tue, Jun 21, 2011 at 10:26 AM, Amol Vedak amol.ve...@mobilewaretech.comwrote: Dear Asterisk Users, I have a Sipura 2000 device, and since last few days I have been searching for its latest firmware for upgrade. Googling tells me that Cisco has stopped the support for this device and I dont have definite idea on where would I be able to find the firmware to upgrade my device. Any help in regards to getting the firmware will be helpful. Regards, Amol -- _ -- 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
