Re: [asterisk-users] Asterisk Support from Digium
Thanks Andrew, But i'm quite confuse with the following: *Q: Does Digium offer SLA guaranteed support for Asterisk?* *A:* Yes. Digium offers SLA guaranteed support, to SLA-entitled customers, for the Certified Asterisk branches. Digium does not offer SLA guaranteed support for other branches or releases. Just for Certify Versions of Asterisk? What does SLA means "exactly"? For example, if i install a FreePBX/Elastix (i'm not a good friend of these systems, but customers always ask for a web interface for management) to a customer, can i buy support from Digium for the Asterisk Release used? It would be nice to now the scope and limits of this support Thanks 2012/11/3 Andrew Latham > On Sat, Nov 3, 2012 at 2:16 PM, Danny Dias > wrote: > > Hello, > > > > I wonder if Digium provides support for Asterisk OpenSource versions as > an > > anual fee or something? > > > > For example, if i download Asterisk 1.8.X (Certified or not...) can i buy > > support from Digium to maintain and help on possible future problems in > my > > configuration? > > > > Thanks > > Yes > > Please review > http://www.digium.com/en/supportcenter/custom-communications-solutions/ > for more information. > > > -- > ~ Andrew "lathama" Latham lath...@gmail.com http://lathama.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 > -- *SIP:* da...@voice.danntel.net <http://www.danntel.net/?page_id=189> *Web: *http://www.danntel.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
[asterisk-users] Asterisk Support from Digium
Hello, I wonder if Digium provides support for Asterisk OpenSource versions as an anual fee or something? For example, if i download Asterisk 1.8.X (Certified or not...) can i buy support from Digium to maintain and help on possible future problems in my configuration? 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] Problems installing DPMA
It wors. Thanks El 12/06/2012 22:21, "Shaun Ruffell" escribió: > On Tue, Jun 12, 2012 at 10:17:46PM +0200, Danny Dias wrote: > > It's weird, already installed avahi with yum install avahi > > > > Now the error is: > > > > [Jun 12 16:13:46] WARNING[19949] loader.c: Error loading module > > 'res_digium_phone.so': /usr/lib/asterisk/modules/res_digium_phone.so: > > undefined symbol: ast_vm_msg_play > > [Jun 12 16:13:46] WARNING[19949] loader.c: Module 'res_digium_phone.so' > > could not be loaded > > > > Something related to voicemail? > > Someone hit this recently on the forums [1]. The fix was to make > sure voicemail was enabled before compiling Asterisk. > > [1] http://forums.asterisk.org/viewtopic.php?f=1&t=82911 > > -- > Shaun Ruffell > Digium, Inc. | Linux Kernel Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems installing DPMA
It's weird, already installed avahi with yum install avahi Now the error is: [Jun 12 16:13:46] WARNING[19949] loader.c: Error loading module 'res_digium_phone.so': /usr/lib/asterisk/modules/res_digium_phone.so: undefined symbol: ast_vm_msg_play [Jun 12 16:13:46] WARNING[19949] loader.c: Module 'res_digium_phone.so' could not be loaded Something related to voicemail? Thanks 2012/6/12 Danny Dias > Thanks Jason, > > I didn't see in any document with an advice of "packages needed". And yes, > i did open a case yesterday, no answer yet! > > BR > > > 2012/6/12 Jason Parker > >> On 06/12/2012 02:56 PM, Danny Dias wrote: >> > Hi, >> > >> > I'm just trying to install the DPMA on my Asterisk. I already made the >> upgrade >> > from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did: >> > >> > /mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185 >> > / >> > *compiling Asterisk-Cert2 1.8.11* >> > /./configure >> > make >> > make install >> > make config >> > / >> > Afther that i register the DPMA license, and finally copied the >> > *res_digium_phone.so* to //usr/lib/asterisk/modules / >> > >> > When i try to load the module on asterisk console this is what i get> >> > >> > /*CLI> module load res_digium_phone.so >> > Unable to load module res_digium_phone.so >> > Command 'module load res_digium_phone.so' failed. / >> > >> > With /tail -f /var/log/asterisk/message / >> > >> > /[Jun 11 18:53:26] WARNING[2554] loader.c: Error loading module >> > 'res_digium_phone.so': libavahi-client.so.3: cannot open shared object >> file: No >> > such file or directory >> > [Jun 11 18:53:26] WARNING[2554] loader.c: Module 'res_digium_phone.so' >> could not >> > be loaded. / >> > >> > Hope you can help >> > >> >> Questions like this should usually be directed to Digium support. >> >> Your issue can be fixed by installing the package containing >> libavahi-client. >> >> On CentOS: yum install avahi >> on Debian/Ubuntu: apt-get install libavahi-client3 >> >> -- >> _ >> -- 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] Problems installing DPMA
Thanks Jason, I didn't see in any document with an advice of "packages needed". And yes, i did open a case yesterday, no answer yet! BR 2012/6/12 Jason Parker > On 06/12/2012 02:56 PM, Danny Dias wrote: > > Hi, > > > > I'm just trying to install the DPMA on my Asterisk. I already made the > upgrade > > from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did: > > > > /mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185 > > / > > *compiling Asterisk-Cert2 1.8.11* > > /./configure > > make > > make install > > make config > > / > > Afther that i register the DPMA license, and finally copied the > > *res_digium_phone.so* to //usr/lib/asterisk/modules / > > > > When i try to load the module on asterisk console this is what i get> > > > > /*CLI> module load res_digium_phone.so > > Unable to load module res_digium_phone.so > > Command 'module load res_digium_phone.so' failed. / > > > > With /tail -f /var/log/asterisk/message / > > > > /[Jun 11 18:53:26] WARNING[2554] loader.c: Error loading module > > 'res_digium_phone.so': libavahi-client.so.3: cannot open shared object > file: No > > such file or directory > > [Jun 11 18:53:26] WARNING[2554] loader.c: Module 'res_digium_phone.so' > could not > > be loaded. / > > > > Hope you can help > > > > Questions like this should usually be directed to Digium support. > > Your issue can be fixed by installing the package containing > libavahi-client. > > On CentOS: yum install avahi > on Debian/Ubuntu: apt-get install libavahi-client3 > > -- > _ > -- 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] Problems installing DPMA
Hi, I'm just trying to install the DPMA on my Asterisk. I already made the upgrade from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did: *mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185 * *compiling Asterisk-Cert2 1.8.11* *./configure make make install make config * Afther that i register the DPMA license, and finally copied the * res_digium_phone.so* to */usr/lib/asterisk/modules * When i try to load the module on asterisk console this is what i get> **CLI> module load res_digium_phone.so Unable to load module res_digium_phone.so Command 'module load res_digium_phone.so' failed. * With *tail -f /var/log/asterisk/message * *[Jun 11 18:53:26] WARNING[2554] loader.c: Error loading module 'res_digium_phone.so': libavahi-client.so.3: cannot open shared object file: No such file or directory [Jun 11 18:53:26] WARNING[2554] loader.c: Module 'res_digium_phone.so' could not be loaded. * Hope you can help -- _ -- 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] Differences between PBX and SBC
That's my question...the sbc provides security over trunking, right? The same can do Asterisk or a Proxy..isn't? Does an SBC can provide any kind of add-value to an Asterisk deployment? El 11/06/2012 20:20, "Paul Belanger" escribió: > On 12-06-11 02:06 PM, Danny Dias wrote: > >> Hello, >> >> I would like to know the difference between encrypt the rtp and signaling >> between two asterisks, or putting an SBC in front of each Asterisk pbx. >> I'm >> trying to understand whether an SBC could fit an Asterisk deployment >> >> What are you expecting the SBC to do? > > -- > Paul Belanger | PolyBeacon, Inc. > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > Github: https://github.com/pabelanger | Twitter: > https://twitter.com/pabelanger > > -- > __**__**_ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/**mailman/listinfo/asterisk-**users<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] Differences between PBX and SBC
Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment 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] Fax Server for Asterisk
Hi Tim, Unfortunately i can't reproduce the scenario because it was a long time ago. But it would be nice to hear from you, what things can be verified within fax and Asterisk? Any TIP on wireshark monitoring? El 31/05/2012 03:08, "Tim Nelson" escribió: > - Original Message - > > I've used Asterisk 1.4.22 with hylafax and iaxmodem and it wasn't > > reliable at all; sometimes the fax reach the destination, sometimes > > not, and even worse, asterisk got froozen...(here using analog lines > > over Sangoma B600 and Digium TDM400P, same behavior with both. > > Other history with same asterisk version but E1 lines, it was > > PERFECT. That's why i ask for analog lines, since not all customers > > has E1. > > Any recommendation/restriction when using hylafax + Asterisk + > > iaxmodem ? > > BR > > If Asterisk was freezing up, that would seem to indicate a problem with > Asterisk, not the Hylafax/IAXmodem components. Of course, details would be > needed to determine why that was the case. > > Regardless, without lockups of Asterisk, reliability of fax is very > dependent on timing and audio quality. Again, details would be needed to > further investigate why you had high failure metrics(specifically your fax > session logs from /var/spool/hylafax/log). > > In general, Hylafax+[1] and IAXmodem is the most rock solid stable fax > solution available, as long as you can get past the initial learning curve. > There is a reason why IAXmodem has not had a release in forever as the > 1.2.0 release is rock solid stable. Hylafax+ continues to be developed with > regular releases, the feature set and functionality are second to none with > hooks for almost any imaginable configuration, and the support via the > mailing lists or available contractors can't be beat. > > > > If you have specifics about your problems with Hylafax and IAXmodem, I'd > love to hear about them to help diagnose, if it is postmortem. > > --Tim > > [1] There *IS* a difference between "Hylafax" (hylafax.org) and > "Hylafax+" (hylafax.sourceforge.net). Please see here: > http://hylafax.sourceforge.net/about.php > > -- > _ > -- 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] Fax Server for Asterisk
Just to clarify, i were using fax machines connected to fxs ports El 30/05/2012 20:31, "Danny Dias" escribió: > I've used Asterisk 1.4.22 with hylafax and iaxmodem and it wasn't reliable > at all; sometimes the fax reach the destination, sometimes not, and even > worse, asterisk got froozen...(here using analog lines over Sangoma B600 > and Digium TDM400P, same behavior with both. > > Other history with same asterisk version but E1 lines, it was PERFECT. > That's why i ask for analog lines, since not all customers has E1. > > Any recommendation/restriction when using hylafax + Asterisk + iaxmodem ? > > BR > El 29/05/2012 22:29, "Carlos Alvarez" escribió: > -- _ -- 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] Fax Server for Asterisk
I've used Asterisk 1.4.22 with hylafax and iaxmodem and it wasn't reliable at all; sometimes the fax reach the destination, sometimes not, and even worse, asterisk got froozen...(here using analog lines over Sangoma B600 and Digium TDM400P, same behavior with both. Other history with same asterisk version but E1 lines, it was PERFECT. That's why i ask for analog lines, since not all customers has E1. Any recommendation/restriction when using hylafax + Asterisk + iaxmodem ? BR El 29/05/2012 22:29, "Carlos Alvarez" escribió: -- _ -- 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] Fax Server for Asterisk
Hi all, Does Hylafax and IAXmodem works with analog lines? or only with E1? I've been checking some commercial solutions (in case Asterisk is not on site, and the customer wants a Fax Server as standalone), i saw FaxBack and Linkcom e-fax But again, if Hylafax and iaxmodem works also with analog lines, that would be better to use. Could you please confirm? any place to check How-To on Hylafax and Iaxmodem? Many thanks!!! 2012/5/29 Carlos Alvarez > > On Tue, May 29, 2012 at 8:03 AM, Warren Selby wrote: > >> On Tue, May 29, 2012 at 3:10 AM, Danny Dias wrote: >> >>> Hello, >>> >>> For those customers with only analog lines, who ask for fax2email and >>> email2fax, whats the most reliable solution available and tested with >>> Asterisk? >>> >>> Thanks >>> >>> >> I've been real happy with using HylaFax+ and Iaxmodem implementations. >> > > > We have a few Hylafax servers in our network. Both it and IAXmodem are a > real bear to learn at first (well, so is Asterisk) but when you get them > working, they are rock solid. I hadn't even thought about it, but it's > been at least a year since I logged into any of our Hylafax servers and did > anything to them. They just work. > > I would estimate I put in a solid 30 hours into learning and configuring > the first server, and then some more time learning additional capabilities > and best practices. But again, since doing that, it's been totally > hands-off. > > I will add though that we also use Fax for Asterisk simply to receive and > turn faxes into PDF for some customers, and that is perfectly stable also. > > > -- > Carlos Alvarez > TelEvolve > 602-889-3003 > > > > -- > _ > -- 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 > -- www.danntel.net *sip:danny4...@thesipschool.com* sip:dann...@opensips.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
[asterisk-users] Fax Server for Asterisk
Hello, For those customers with only analog lines, who ask for fax2email and email2fax, whats the most reliable solution available and tested with Asterisk? 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] Deleting OLD Voicemails
I did not understand. What do you mean with renumber all the messages? El 25/05/2012 02:27, "Edwin Lam" escribió: > On 5/23/12 2:42 AM, Danny Dias wrote: > >> Can i delete like this: >> >> rm -rf /var/spool/asterisk/voicemail/**voicemailcontextcustomer/300/** >> INBOX/*.* >> >> Is that ok? will this break something? >> > > that's ok > no it shouldn't break anything. > however if you're going to delete some of the messages. you have to > renumber all the messages so that they are consecutive otherwise > the voicemail application may give you grief. > > A little doubt here, once the user hears the voicemail using the phone, >> the >> message is automatically moved to Old folder, is that right? >> > > yes > > > -- > Edwin Lam > Systems Engineer, OfficeWyze, Inc. > Ph: +1 415 439 4988 Fax: +1 415 283 3370 > http://pgpkeys.mit.edu:11371/**pks/lookup?op=get&search=**0xD6506D20<http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20> > > > -- > __**__**_ > -- 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<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] Deleting OLD Voicemails
Hi, thanks for your answers... Can i delete like this: rm -rf /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.* Is that ok? will this break something? A little doubt here, once the user hears the voicemail using the phone, the message is automatically moved to Old folder, is that right? Many thanks! 2012/5/23 Mehmet Avcioglu > > You can delete old files, it won't break anything. Also to prevent saving > files in multiple formats, edit voicemail.conf and change format parameter > under general. > > -- > Mehmet Avcioglu > meh...@activecom.net > > On May 23, 2012, at 1:03 AM, Danny Dias wrote: > > Thanks Jason, > > But how to delete them? there are a lot of old voicemails, but i don't > want to break the app_voicemail. > > > > 2012/5/22 Jason Parker > >> On 05/22/2012 04:54 PM, Danny Dias wrote: >> > There are 4 files for each voicemail: >> > >> > msg.gsm >> > msg.txt >> > msg.wav >> > msg.WAV >> > >> >> That is perfectly normal. The .txt file is metadata that contains things >> like >> caller ID and duration. Asterisk will also save voicemails into every >> format >> you have specified in voicemail.conf. >> >> -- >> _ >> -- 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 >> > > > > -- > www.danntel.net > *sip:danny4...@thesipschool.com* > sip:dann...@opensips.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 > -- www.danntel.net *sip:danny4...@thesipschool.com* sip:dann...@opensips.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] Deleting OLD Voicemails
Thanks Jason, But how to delete them? there are a lot of old voicemails, but i don't want to break the app_voicemail. 2012/5/22 Jason Parker > On 05/22/2012 04:54 PM, Danny Dias wrote: > > There are 4 files for each voicemail: > > > > msg.gsm > > msg.txt > > msg.wav > > msg.WAV > > > > That is perfectly normal. The .txt file is metadata that contains things > like > caller ID and duration. Asterisk will also save voicemails into every > format > you have specified in voicemail.conf. > > -- > _ > -- 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 > -- www.danntel.net *sip:danny4...@thesipschool.com* sip:dann...@opensips.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
[asterisk-users] Deleting OLD Voicemails
Hello, I was checking how to DELETE old voicemail from Asterisk, for my extension 300, i have 20 MB [root@pbx INBOX]# pwd /var/spool/asterisk/voicemail/default/300/INBOX [root@pbx INBOX]# du -s -h 20M There are 4 files for each voicemail: msg.gsm msg.txt msg.wav msg.WAV I've read on some forums that deleting could break the app_voicemail; so i'm a little afraid of break something. Why 4 files for each voicemail? Is there any "best way to do" without breaking app_voicemail? Many thanks -- www.danntel.net *sip:danny4...@thesipschool.com* sip:dann...@opensips.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] DPMA for Digium Phones
By the way, the DPMA is only available for Asterisk Certified, is that right? is there any problem for an Asterisk production server on a customer with Asterisk OpenSource 1.8.5 to migrate to Asterisk Certified? What is exactly an Asterisk Certified? Do we have to pay for some license? Thanks 2012/5/21 Danny Dias > Hello, > > I have a question regarding DPMA for Digium Phones, if i install the DPMA > on my Asterisk Server "A", and then, i move the phone to register into > another Asterisk Server "B", can i install for "free" another DPMA license > for my digium phones on this second server? can i move the DPMA from one > server to another or i have to buy new licenses? > > Thanks > > -- > www.danntel.net > *sip:danny4...@thesipschool.com* > sip:dann...@opensips.org > > > > > -- www.danntel.net *sip:danny4...@thesipschool.com* sip:dann...@opensips.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
[asterisk-users] DPMA for Digium Phones
Hello, I have a question regarding DPMA for Digium Phones, if i install the DPMA on my Asterisk Server "A", and then, i move the phone to register into another Asterisk Server "B", can i install for "free" another DPMA license for my digium phones on this second server? can i move the DPMA from one server to another or i have to buy new licenses? Thanks -- www.danntel.net *sip:danny4...@thesipschool.com* sip:dann...@opensips.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] R-Series with NON-DIGIUM card on servers
Thanks Kevin. Buying one for Spain right now ;) 2012/5/15 Kevin P. Fleming > On 05/12/2012 12:07 PM, Danny Dias wrote: > > What about the Database and recording calls replication? as i could see, >> the RSeries does not take into account these data. >> > > The Digium R-series devices are electronic switches used for routing > telephony circuits; they don't have any part in the actual failover > process, data replication, or anything of the sort. All of those functions > need to be handled via software on the server(s) involved. The R-series > user's manual describes one way this can be done using Asterisk and open > source tools commonly available on Linux distributions. > > > -- > 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<http://lists.digium.com/mailman/listinfo/asterisk-users> > -- www.danntel.net *sip:danny4...@thesipschool.com* sip:dann...@opensips.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] R-Series with NON-DIGIUM card on servers
Thanks, What about the Database and recording calls replication? as i could see, the RSeries does not take into account these data. Thanks 2012/5/12 Kevin P. Fleming > On 05/11/2012 10:46 PM, Danny Dias wrote: > >> Hi, >> >> I would like to know if the servers (A and B) could use boards >> non-digium with the R-Series HA product from Digium, i have a couple of >> B600E Sangoma to put on each server and use the R-series to provide HA. >> Is that possible? >> > > Yes. The Digium R-series failover appliances will work with any device > that uses the appropriate type of PSTN circuits (digital or analog, > depending on the R-series model), even a legacy PBX. > > -- > 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<http://lists.digium.com/mailman/listinfo/asterisk-users> > -- www.danntel.net *sip:danny4...@thesipschool.com* sip:dann...@opensips.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] Digium IP Phones
Does the D40 will support the option to develope apps? As i could see on videos only the D70 has the apps button, and also, the lcd screen is smaller. Right? Enviado desde mi Samsung Galaxy S II El 10/05/2012 12:44, "Kevin P. Fleming" escribió: > On 05/09/2012 08:38 PM, Danny Dias wrote: > >> Hello, >> >> Im looking to buy a digium phone D70 unit just for testing on lab; to >> really understand the phone and features. >> >> I cant find any website with opinions; any here? Are they really >> valuable to the price? (D70 quite expensive) >> >> Does the SDK for building apps is usable? Can you build powerfull apps? >> Examples? >> > > The phone app SDK has not been released yet, it's still under development. > > -- > 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<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] R-Series with NON-DIGIUM card on servers
Hi, I would like to know if the servers (A and B) could use boards non-digium with the R-Series HA product from Digium, i have a couple of B600E Sangoma to put on each server and use the R-series to provide HA. Is that possible? Thanks -- www.danntel.net *sip:danny4...@thesipschool.com* sip:dann...@opensips.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
[asterisk-users] Digium IP Phones
Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull apps? Examples? Many 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] Asterisk + Phones behind different Nat Firewalls
Thanks, But if i open rtp ports from 1-2 how would you ping ports from both sides to not loose rtp or having one way audio if the ports are choosen randomly between 10.000-20.000 in every call? The keep alive works for signalling (Asterisks sends Options to the contact), but not for RTP. For RTP i think it is mandatory to have an STUN server ir RTP proxy. Right? El 27/04/2012 12:15, escribió: > The asterisk side has to have the router ports 5060 and 1-2 > forwarded to asterisk these are the standard ports but you could cut way > down on the rtp ports in rtp.conf then you have to tell asterisk what's > the external ip of your nat and most of the times this should work today no > problem lots of us here have it working that way (of course you have to > take care of security fail2ban etc ) > On the phone side you might have to use stun but it depends on the > firewall also you should set the phone to send a nat keep alive each 30 > seconds (asterisk also sends a options packet to keep the nat open but > doesn't always work ok ) > > -Original Message- > From: Danny Dias > Sender: asterisk-users-boun...@lists.digium.com > Date: Fri, 27 Apr 2012 10:22:38 > To: Asterisk Users Mailing List - Non-Commercial Discussion< > asterisk-users@lists.digium.com> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Asterisk + Phones behind different Nat >Firewalls > > -- > _ > -- 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] POTS(FXO) line getting Red alarm after first ring(5 or 6 seconds)
Btw, red alarms means phisical problemscheck cable first.
El 27/04/2012 10:23, "Danny Dias" escribió:
> Did you asked OpenVOX for support?
> El 27/04/2012 01:48, "John Millican" escribió:
>
>> Hello,
>> I have an OpenVox A400E02 (2FXO) in a box running Debian 6.0.2 running
>> Asterisk 1.8.6.0. I have to POTS line on it from Verizon in Virginia, USA.
>> Whenever I place a call to one of the two lines I get a red alam and then
>> it clears and repeats this till I hang up. There is no caller ID on the
>> Line (boss won't pay for it).
>> Any help is most appreciated.
>> TIA,
>> JohnM
>>
>> lspci relevent output:
>> 08:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
>> interface
>>
>> cat /proc/interrupts:
>> 19: 54390 1286431613 IO-APIC-fasteoi wctdm
>> (no shared interupts)
>>
>> dahdi show channels
>> Chan Extension Context Language MOH InterpretBlocked
>>State
>> pseudodefaultdefault
>> In Service
>> 1altrurstn default
>> In Service
>> 2altrurstn default
>> In Service
>>
>> PBX1:/home/jmillican# dahdi_cfg -vvv
>> DAHDI Tools Version - 2.5.0.1
>>
>> DAHDI Version: 2.5.0.1
>> Echo Canceller(s): HWEC, MG2
>> Configuration
>> ==
>>
>> Channel map:
>>
>> Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
>> Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
>>
>> 2 channels to configure.
>>
>> Setting echocan for channel 1 to mg2
>> Setting echocan for channel 2 to mg2
>>
>>
>> in chan_dahdi.conf
>> [channels]
>> context=altrurstn
>> signalling=fxs_ks
>> rxwink=300
>> usecallerid=yes
>> hidecallerid=no
>> callwaiting=yes
>> usecallingpres=yes
>> callwaitingcallerid=yes
>> threewaycalling=yes
>> transfer=yes
>> canpark=yes
>> cancallforward=yes
>> callreturn=yes
>> echocancel=yes
>> echocancelwhenbridged=no
>> faxdetect=incoming
>> echotraining=800
>> rxgain=0.0
>> txgain=0.0
>> callgroup=1
>> pickupgroup=1
>>
>> immediate=no
>>
>> #include dahdi_additional.conf
>> #include dahdi-channels.conf
>>
>> in dahdi-channels.conf
>> ; Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
>> ;;; line="1 WCTDM/4/0"
>> signalling=fxs_ks
>> callerid=asreceived
>> group=0
>> context=altrurstn
>> channel => 1
>> callerid=
>> group=
>> context=altrurstn
>>
>> ;;; line="2 WCTDM/4/1"
>> signalling=fxs_ks
>> callerid=asreceived
>> group=0
>> context=altrurstn
>> channel => 2
>> callerid=
>> group=
>> context=altrurstn
>>
>> /etc/dahdi/modules loads only wctdm.
>>
>> /etc/dahdi/system.conf:
>> fxsks=1
>> echocanceller=mg2,1
>> fxsks=2
>> echocanceller=mg2,2
>>
>> Relevent Extensions.conf:
>> [altrurstn-in]
>>
>> exten => s,1,Wait(1);
>> exten => s,n,Set(CDR(accountcode)=**fromoustide)
>> exten => s,n,Set(CDR(userfield)=POTS-${**EXTEN})
>> exten => s,n,GoTo(999,1);
>>
>> exten => 999,1,Answer();
>> exten => 999,n,NoOp(${CALLERID(all)});
>> exten => 999,n,wait(1);
>> exten => 999,n,Set(foo=0);
>> exten => 999,n,Set(count=0);
>> exten => 999,n,Read(foo,0001&0002,4,,,**2);
>> exten => 999,n,GoToIf($["${foo}"="9"]?**directory);
>> exten => 999,n,GoToIf($["${foo}"="0"]?**oper)
>> exten => 999,n,GoToIf($["${LEN(${foo})}**" <
>> "4"]?restart:altrurstn,${foo},**1);
>> exten => 999,n(restart),Set(COUNT=$[${**COUNT} + 1]);
>> exten => 999,n,NoOp(${COUNT});
>> exten => 999,n,GoToIf($["${COUNT}" > "1"]?oper:continue);
>> exten => 999,n(continue),Read(foo,0002,**4,,,2);
>> exten => 999,n,GoToIf($["${foo}"="9"]?**directory);
>> exten => 999,n,GoToIf($["${foo}"="0"]?**oper)
>> exten => 999,n,GoToIf($["${LEN(${foo})}**"<"4"]?restart:altrurstn,${**
>> foo},1);
>> exten => 999,n(oper),GoTo(0,1);
>> exten => 999,n(directory),Directory(**default,altrurstn,p(500));
>> exten => 999,n,Hangup();
>>
>> What I get in the CLI:
&
Re: [asterisk-users] POTS(FXO) line getting Red alarm after first ring(5 or 6 seconds)
Did you asked OpenVOX for support?
El 27/04/2012 01:48, "John Millican" escribió:
> Hello,
> I have an OpenVox A400E02 (2FXO) in a box running Debian 6.0.2 running
> Asterisk 1.8.6.0. I have to POTS line on it from Verizon in Virginia, USA.
> Whenever I place a call to one of the two lines I get a red alam and then
> it clears and repeats this till I hang up. There is no caller ID on the
> Line (boss won't pay for it).
> Any help is most appreciated.
> TIA,
> JohnM
>
> lspci relevent output:
> 08:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
> interface
>
> cat /proc/interrupts:
> 19: 54390 1286431613 IO-APIC-fasteoi wctdm
> (no shared interupts)
>
> dahdi show channels
> Chan Extension Context Language MOH InterpretBlocked
>State
> pseudodefaultdefault
> In Service
> 1altrurstn default
> In Service
> 2altrurstn default
> In Service
>
> PBX1:/home/jmillican# dahdi_cfg -vvv
> DAHDI Tools Version - 2.5.0.1
>
> DAHDI Version: 2.5.0.1
> Echo Canceller(s): HWEC, MG2
> Configuration
> ==
>
> Channel map:
>
> Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
> Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
>
> 2 channels to configure.
>
> Setting echocan for channel 1 to mg2
> Setting echocan for channel 2 to mg2
>
>
> in chan_dahdi.conf
> [channels]
> context=altrurstn
> signalling=fxs_ks
> rxwink=300
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=no
> faxdetect=incoming
> echotraining=800
> rxgain=0.0
> txgain=0.0
> callgroup=1
> pickupgroup=1
>
> immediate=no
>
> #include dahdi_additional.conf
> #include dahdi-channels.conf
>
> in dahdi-channels.conf
> ; Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
> ;;; line="1 WCTDM/4/0"
> signalling=fxs_ks
> callerid=asreceived
> group=0
> context=altrurstn
> channel => 1
> callerid=
> group=
> context=altrurstn
>
> ;;; line="2 WCTDM/4/1"
> signalling=fxs_ks
> callerid=asreceived
> group=0
> context=altrurstn
> channel => 2
> callerid=
> group=
> context=altrurstn
>
> /etc/dahdi/modules loads only wctdm.
>
> /etc/dahdi/system.conf:
> fxsks=1
> echocanceller=mg2,1
> fxsks=2
> echocanceller=mg2,2
>
> Relevent Extensions.conf:
> [altrurstn-in]
>
> exten => s,1,Wait(1);
> exten => s,n,Set(CDR(accountcode)=**fromoustide)
> exten => s,n,Set(CDR(userfield)=POTS-${**EXTEN})
> exten => s,n,GoTo(999,1);
>
> exten => 999,1,Answer();
> exten => 999,n,NoOp(${CALLERID(all)});
> exten => 999,n,wait(1);
> exten => 999,n,Set(foo=0);
> exten => 999,n,Set(count=0);
> exten => 999,n,Read(foo,0001&0002,4,,,**2);
> exten => 999,n,GoToIf($["${foo}"="9"]?**directory);
> exten => 999,n,GoToIf($["${foo}"="0"]?**oper)
> exten => 999,n,GoToIf($["${LEN(${foo})}**" <
> "4"]?restart:altrurstn,${foo},**1);
> exten => 999,n(restart),Set(COUNT=$[${**COUNT} + 1]);
> exten => 999,n,NoOp(${COUNT});
> exten => 999,n,GoToIf($["${COUNT}" > "1"]?oper:continue);
> exten => 999,n(continue),Read(foo,0002,**4,,,2);
> exten => 999,n,GoToIf($["${foo}"="9"]?**directory);
> exten => 999,n,GoToIf($["${foo}"="0"]?**oper)
> exten => 999,n,GoToIf($["${LEN(${foo})}**"<"4"]?restart:altrurstn,${**
> foo},1);
> exten => 999,n(oper),GoTo(0,1);
> exten => 999,n(directory),Directory(**default,altrurstn,p(500));
> exten => 999,n,Hangup();
>
> What I get in the CLI:
> [Apr 26 19:26:53] -- Starting simple switch on 'DAHDI/1-1'
> [Apr 26 19:26:53] -- Executing [s@altrurstn-in:1] Wait("DAHDI/1-1",
> "1") in new stack
> [Apr 26 19:26:54] -- Executing [s@altrurstn-in:2] Set("DAHDI/1-1",
> "CDR(accountcode)=fromoustide"**) in new stack
> [Apr 26 19:26:54] -- Executing [s@altrurstn-in:3] Set("DAHDI/1-1",
> "CDR(userfield)=POTS-s") in new stack
> [Apr 26 19:26:54] -- Executing [s@altrurstn-in:4] Goto("DAHDI/1-1",
> "999,1") in new stack
> [Apr 26 19:26:54] -- Goto (altrurstn-in,999,1)
> [Apr 26 19:26:54] -- Executing [999@altrurstn-in:1]
> Answer("DAHDI/1-1", "") in new stack
> [Apr 26 19:26:54] -- Executing [999@altrurstn-in:2] NoOp("DAHDI/1-1",
> """ <>") in new stack
> [Apr 26 19:26:54] -- Executing [999@altrurstn-in:3] Wait("DAHDI/1-1",
> "1") in new stack
> [Apr 26 19:26:55] WARNING[11189]: chan_dahdi.c:7728 handle_alarms:
> Detected alarm on channel 1: Red Alarm
> [Apr 26 19:26:55] == Spawn extension (altrurstn-in, 999, 3) exited
> non-zero on 'DAHDI/1-1'
> [Apr 26 19:26:55] -- Hanging up on 'DAHDI/1-1'
> [Apr 26 19:26:55] -- Hungup 'DAHDI/1-1'
> [Apr 26 19:26:58] NOTICE[11159]: sig_analog.c:3709
> analog_handle_init_event: Alarm cleared on channel 1
> [Apr 26 19:26:59] -- Starting simple switch on 'DAHDI/1-1'
> [Apr 26 19:26:59] -- Executing [s@altrurstn-in:1
Re: [asterisk-users] Asterisk + Phones behind different Nat Firewalls
Ok understood. The signaling wont be a problem, but not the same with rtp as it uses randomly ports. The idea is to have an intermediary who could delivers both ports and ping them to both sides to keep nating open on routers, this is what i do with rtp proxy within opensips. But in this case no OpenSIPS. The router are Comtrend and linksys. Are you sure that works for you with the same environment as mine? Its just that im trying to understand it technically (sdp headers + sip headers) and i do not understand how the rtp will reach both phones on different nat sides The routers does not have ALG. El 26/04/2012 23:19, escribió: > Well you have to tell asterisk what's the external ip of the nat else its > never gone work > Look at externip and localnet > > -Original Message- > From: Carlos Alvarez > Sender: asterisk-users-boun...@lists.digium.com > Date: Thu, 26 Apr 2012 14:15:39 > To: Asterisk Users Mailing List - Non-Commercial Discussion< > asterisk-users@lists.digium.com> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Asterisk + Phones behind different Nat >Firewalls > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Phones behind different Nat Firewalls
Does not work for me! El 26/04/2012 20:14, "Carlos Alvarez" escribió: > > > On Thu, Apr 26, 2012 at 10:47 AM, Danny Dias wrote: > >> I cant put public ip adress to the asterisk server. >> >> The main problem i see is with the sip headers (contact, sdp ip and >> ports, etc)...with the result of one way audio or no registrstion at all. >> Does an STUN server should works? >> AGAIN: phones in one site behind nat and PBX in another site also behind >> nat >> > > No problem. I do that all the time. That's what I said in my message. > > -- > Carlos Alvarez > TelEvolve > 602-889-3003 > > > > -- > _ > -- 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 + Phones behind different Nat Firewalls
I cant put public ip adress to the asterisk server. The main problem i see is with the sip headers (contact, sdp ip and ports, etc)...with the result of one way audio or no registrstion at all. Does an STUN server should works? AGAIN: phones in one site behind nat and PBX in another site also behind nat El 26/04/2012 19:31, "Carlos Alvarez" escribió: > > > On Thu, Apr 26, 2012 at 9:54 AM, Danny Dias wrote: > >> I have a doubt (basic i guess, but not for me). I have an escenario where >> customer site has Asterisk PBX behind Nat/firewall with private IP address >> and sone phones also; BUT there are some other phones on different sites >> and of course behind its nat/firewalls; with IAX i have no problem, but >> customer wants to use SIP phones and there is no way to put IP public >> address for the Asterisk Server. >> > > When you say you can't give the server a public IP, do you mean you can't > put it outside of NAT, or you can't use a public IP at all? With most > routers, NAT from a public IP to a private one for the server works just > fine. I've used Cisco, Juniper, and Sonicwall in front of an Asterisk > server, NAT IP, and SIP with no problems. > > > -- > Carlos Alvarez > TelEvolve > 602-889-3003 > > > > -- > _ > -- 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] Asterisk + Phones behind different Nat Firewalls
Hello, I have a doubt (basic i guess, but not for me). I have an escenario where customer site has Asterisk PBX behind Nat/firewall with private IP address and sone phones also; BUT there are some other phones on different sites and of course behind its nat/firewalls; with IAX i have no problem, but customer wants to use SIP phones and there is no way to put IP public address for the Asterisk Server. What do you recommend? Any advice? Many thanks in advance -- _ -- 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] Calling a group of phones and force the speaker
Hi, I wonder, if there is a way to call from A phone to a group of phones (B, C and D) and force these phones to activate automatically the speaker Is that possible? Many thanks in advance -- www.danntel.net *sip:danny4...@thesipschool.com* sip:dann...@opensips.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] res_musiconhold.c Bug - Patch to solve?
2010/11/22 John Novack > > > Danny Dias wrote: > > Hello John, > > What i am asking is if i can apply this patch manually or something like > this without making any upgrade of Asterisk, has anyone done this before? > > I can't answer that question. > > ummm why not? is something wrong? > Or i have to upgrade my Asterisk versions...i don't really want to do > this... > > Why not? MANY fixes have been included in the upgrades. > Improved security at the least. There are 10-15 versions between where you > are operating and what is current > > I'm sure that the upgrade will fix this, but if applying the patch without making any upgrade will be better for me, my asterisk servers are working with many calls, realtime, fop etc...and an upgrade could make something happen... > John Novack > > > Thanks in Advance! > > 2010/11/22 John Novack > >> Hasn't this been fixed in later versions? >> 1.4.37 is current, or at least it was in the last few days. >> >> Upgrading with no reason isn't suggested, but in this case you have a good >> reason, and if you dig deep enough you may find the fix is already in place. >> >> John Novack >> >> >> >> Danny Dias wrote: >> >>> Hello Asterisk community, >>> >>> We are having some problems with crashes in Asterisk, my asterisk >>> versions are 1.4.24.1 and 1.4.23.2. I have found this: >>> >>> "~/work/asterisk-branch-1.4$ svn log -c 260345 >>> >>> r260345 | mmichelson | 2010-04-30 22:08:15 +0200 (Fri, 30 Apr 2010) | 18 >>> lines >>> >>> Fix potential crash from race condition due to accessing channel data >>> without the channel locked. >>> >>> In res_musiconhold.c, there are several places where a channel's >>> stream's existence is checked prior to calling ast_closestream on it. The >>> issue >>> here is that in several cases, the channel was not locked while checking >>> the >>> stream. The result was that if two threads checked the state of the >>> channel's >>> stream at approximately the same time, then there could be a situation >>> where >>> both threads attempt to call ast_closestream on the channel's stream. The >>> result >>> here is that the refcount for the stream would go below 0, resulting in a >>> crash. >>> >>> I have added proper channel locking to res_musiconhold.c to ensure that >>> we do not try to check chan->stream without the channel locked. A >>> Digium customer has been using this patch for several weeks and has not >>> had any crashes since applying the patch. >>> >>> ABE-2147 >>> " >>> >>> How can i apply this patch on my asterisk versions: 1.4.24.1 and >>> 1.4.23.2? do i have to apply this patch manually? >>> >>> Thanks in advance for your help >>> >>> >>> >> >> -- >> >> Dog is my Co-pilot >> >> > > > -- > Ing. Danny Dias > www.DannTEL.net > > > -- > > Dog is my Co-pilot > > -- Ing. Danny Dias www.DannTEL.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] res_musiconhold.c Bug - Patch to solve?
Hello John, What i am asking is if i can apply this patch manually or something like this without making any upgrade of Asterisk, has anyone done this before? Or i have to upgrade my Asterisk versions...i don't really want to do this... Thanks in Advance! 2010/11/22 John Novack > Hasn't this been fixed in later versions? > 1.4.37 is current, or at least it was in the last few days. > > Upgrading with no reason isn't suggested, but in this case you have a good > reason, and if you dig deep enough you may find the fix is already in place. > > John Novack > > > > Danny Dias wrote: > >> Hello Asterisk community, >> >> We are having some problems with crashes in Asterisk, my asterisk >> versions are 1.4.24.1 and 1.4.23.2. I have found this: >> >> "~/work/asterisk-branch-1.4$ svn log -c 260345 >> >> r260345 | mmichelson | 2010-04-30 22:08:15 +0200 (Fri, 30 Apr 2010) | 18 >> lines >> >> Fix potential crash from race condition due to accessing channel data >> without the channel locked. >> >> In res_musiconhold.c, there are several places where a channel's >> stream's existence is checked prior to calling ast_closestream on it. The >> issue >> here is that in several cases, the channel was not locked while checking >> the >> stream. The result was that if two threads checked the state of the >> channel's >> stream at approximately the same time, then there could be a situation >> where >> both threads attempt to call ast_closestream on the channel's stream. The >> result >> here is that the refcount for the stream would go below 0, resulting in a >> crash. >> >> I have added proper channel locking to res_musiconhold.c to ensure that >> we do not try to check chan->stream without the channel locked. A >> Digium customer has been using this patch for several weeks and has not >> had any crashes since applying the patch. >> >> ABE-2147 >> " >> >> How can i apply this patch on my asterisk versions: 1.4.24.1 and >> 1.4.23.2? do i have to apply this patch manually? >> >> Thanks in advance for your help >> >> >> > > -- > > Dog is my Co-pilot > > -- Ing. Danny Dias www.DannTEL.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
[asterisk-users] res_musiconhold.c Bug - Patch to solve?
Hello Asterisk community, We are having some problems with crashes in Asterisk, my asterisk versions are 1.4.24.1 and 1.4.23.2. I have found this: "~/work/asterisk-branch-1.4$ svn log -c 260345 r260345 | mmichelson | 2010-04-30 22:08:15 +0200 (Fri, 30 Apr 2010) | 18 lines Fix potential crash from race condition due to accessing channel data without the channel locked. In res_musiconhold.c, there are several places where a channel's stream's existence is checked prior to calling ast_closestream on it. The issue here is that in several cases, the channel was not locked while checking the stream. The result was that if two threads checked the state of the channel's stream at approximately the same time, then there could be a situation where both threads attempt to call ast_closestream on the channel's stream. The result here is that the refcount for the stream would go below 0, resulting in a crash. I have added proper channel locking to res_musiconhold.c to ensure that we do not try to check chan->stream without the channel locked. A Digium customer has been using this patch for several weeks and has not had any crashes since applying the patch. ABE-2147 " How can i apply this patch on my asterisk versions: 1.4.24.1 and 1.4.23.2? do i have to apply this patch manually? Thanks in advance for your help -- Ing. Danny Dias www.DannTEL.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
[asterisk-users] dialing from asterisk console?
Hello friends, I'm trying to make a simple call from asterisk CLI, but is quite confuse i followed the information here: http://www.voip-info.org/wiki/view/Asterisk+CLI+dial and changed my extensions.conf like this: alsa.conf [general] autoanswer=no context=consolecontext extension=100 By the way, how do i know if my console is using the channel driver ALSA or OSS? then, in extensions.conf: [consolecontext] exten => 100,1,Dial($DEMO) And then, from the Asterisk CLI: many attempts: CLI> dial No such extension 's' in context 'default' CLI> dial 100 No such extension '100' in context 'default' What am i doing wrong? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP 401
By the way, Could you please make a "better picture" of your work? try using insecure=invite,port, that's the key! by the way, try to use IPs rather than domain names. And check here also: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf register => user[:secret[:authuse...@host[:port][/extension] 2010/10/20 Danny Dias > Zakir, > > Have you checked the RFC3261? > > 21.4.2 401 Unauthorized > The request requires user authentication. This response is issued by > UASs and registrars, while 407 (Proxy Authentication Required) is > used by proxy servers. > > > > 2010/10/20 Zakir Mahomedy > >> Hi >> >> >> >> I am trying to get 2 accounts from voipblaster to talk to each other. >> >> Calls withing voipblaster network is free. If I configure two sip >> clients with the two accounts it works fine >> >> however with Asterisk I am getting SIP 401 >> >> >> >> In my Sip.conf file I under general >> >> >> >> register = >> user:passw...@sip.voipblaster.com >> >> >> >> then I have a sip peer >> >> >> >> >> >> [FreeCall](default) >> type= friend >> context= incoming >> username = kiks2010 >> secret = password >> host= sip.voipblast.com >> fromuser = kiks2010 >> fromdomain = sip.voipblast.com >> insecure=very >> qualify=yes >> >> >> >> these are the sip debug logs >> >> >> >> v=0 >> o=kiks2010 1287592622 1287592622 IN IP4 77.72.168.99 >> s=SIP Call >> c=IN IP4 77.72.168.99 >> t=0 0 >> m=audio 11538 RTP/AVP 8 101<-> >> >> >> --- (11 headers 9 lines) --- >> == Using SIP RTP CoS mark 5 >> Sending to 77.72.174.128 : 5060 (NAT) >> Using INVITE request as basis request - >> 64de05c42e7b4ef2a0678f999c0ed...@77.72.174.128 >> Found peer 'FreeCall' for 'ajs2010' from 77.72.174.128:5060 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=ptime:20 >> >> >> >> <--- Reliably Transmitting (NAT) to 77.72.174.128:5060 ---> >> SIP/2.0 401 Unauthorized >> Via: SIP/2.0/UDP 77.72.174.128:5060 >> ;branch=z9hG4bK6ff0e241f3fd4d0b9c137d616de1fe1f;received=77.72.174.128 >> From: "ajs2010" > >;tag=330113ac4c51ef02d4ef70 >> >> >> >> Any help info will be appreciated >> >> thanks >> >> >> >> Zakir >> >> >> >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP 401
Zakir, Have you checked the RFC3261? 21.4.2 401 Unauthorized The request requires user authentication. This response is issued by UASs and registrars, while 407 (Proxy Authentication Required) is used by proxy servers. 2010/10/20 Zakir Mahomedy > Hi > > > > I am trying to get 2 accounts from voipblaster to talk to each other. > > Calls withing voipblaster network is free. If I configure two sip > clients with the two accounts it works fine > > however with Asterisk I am getting SIP 401 > > > > In my Sip.conf file I under general > > > > register = > user:passw...@sip.voipblaster.com > > > > then I have a sip peer > > > > > > [FreeCall](default) > type= friend > context= incoming > username = kiks2010 > secret = password > host= sip.voipblast.com > fromuser = kiks2010 > fromdomain = sip.voipblast.com > insecure=very > qualify=yes > > > > these are the sip debug logs > > > > v=0 > o=kiks2010 1287592622 1287592622 IN IP4 77.72.168.99 > s=SIP Call > c=IN IP4 77.72.168.99 > t=0 0 > m=audio 11538 RTP/AVP 8 101<-> > > > --- (11 headers 9 lines) --- > == Using SIP RTP CoS mark 5 > Sending to 77.72.174.128 : 5060 (NAT) > Using INVITE request as basis request - > 64de05c42e7b4ef2a0678f999c0ed...@77.72.174.128 > Found peer 'FreeCall' for 'ajs2010' from 77.72.174.128:5060 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > > > > <--- Reliably Transmitting (NAT) to 77.72.174.128:5060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 77.72.174.128:5060 > ;branch=z9hG4bK6ff0e241f3fd4d0b9c137d616de1fe1f;received=77.72.174.128 > From: "ajs2010" >;tag=330113ac4c51ef02d4ef70 > > > > Any help info will be appreciated > > thanks > > > > Zakir > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] checking CDR
Hello Asterisk Community, Is there a way to check in asterisk cdrs and extension forwarded? I mean, i'm calling to a ISDN number, wich goes to extension 8222, but this extension is forwarded to another one, the problem is that in CDRs i am able to see the the first step of the call, but never see the forwarded extension, how can i do that? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces
Thanks Steve, I got the picture :) THANK!!! But my doubt is about the cable, what cable should i use? i have a Sangoma A108D in one machine (one machine with one card). What cable should i do? how can i make it? Best Regards! 2010/10/5 Steve Murphy > > > On Tue, Oct 5, 2010 at 1:02 PM, Danny Dias wrote: > >> Hello my friend Ingmar, >> >> I would like to know the cable you used? how was the connection? i'm using >> this one: >> >> http://wiki.sangoma.com/Pinouts#A108 Loop Back >> >> Is this ok? what should i do my friend, my problems are "understand" the >> fisicall connection :( >> >> Best Regards!!! >> >> 2010/9/24 Ingmar Steen >> >>> Hi DD, >>> >>> >>> >>> We usually use loopback cables and use the open source SIP test tool >>> “SIPp” to initiate SIP calls that are sent from one group of 4 ports to >>> another group of 4 ports. >>> >>> >>> >>> Met vriendelijke groet, >>> >>> Ingmar Steen >>> >>> Teleknowledge >>> >>> >>> >>> *Van:* asterisk-users-boun...@lists.digium.com [mailto: >>> asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias >>> *Verzonden:* vrijdag 24 september 2010 11:05 >>> *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion >>> *Onderwerp:* [asterisk-users] How to test BIG traffic through >>> DAHDI/WANPIPEinterfaces >>> >>> >>> >>> Hello Community, >>> >>> >>> >>> I need to test or simulate many calls through dahdi/wanpipe, i have a >>> Sangoma A108D, and i need to test the stability of the >>> card/drivers/firmwares with a test environment, do you think is possible? >>> >>> >>> >>> What should i do? using some loopback cable maybe? >>> >>> >>> >>> Thanks in advance >>> >>> >>> >>> DD >>> >> > I set up two machines with T1 interfaces, and connected the two with an > appropriate t1 cable. > One was acting as a network (master), the other as a subscriber (slave) > (for timing). wrote two dialplans, one for each machine, > that would answer an incoming call on one dahdi line, and call to the next > numbered line on the other > machine. The other machine was similarly outfit. I'd define the extension > for the first line on the t1, > and call it with any phone you desire. That call will cascade into 23 > separate interlinked calls. If you are > clever, the last call in should dial another real phone you have on-hand. > > You get the picture... right? Phone A dials the exten to call the first > exten on the other machine. The > dialplan should use the first channel on the t1 to place a call to the > first exten on the other machine. > On the other machine, the incoming call on channel 1 is answered, and then > a dial to the second extension > on the first machine, over the 2nd t1 channel. The first machine answers, > and uses the 3rd channel > to call the other machine and so on till all channels are being used. > The last exten answers and dials > a phone (dahdi or SIP, no matter) that you pick up. Such a looped call > should probably be awful, but > it's going thru 23 t1 channels! > > If you have two t1 interaces in a single card (or two cards), then you do > this on one machine. > > Another approach: set up equal numbers of FZO and FXS lines, and similarly > loop s single call thru all the > channels.This would require just one machine. > > Other approaches would involve running multiple threads to call an > extension and then hang up and > repeating this over and over again on all channels to ascertain the load > placed just by call setup and tear-down. > This kind of load is different than when all lines are just shoveling data > back and forth. > > Watch your load averages, your %cpu, your swap, etc, as the tests are in > full swing. > > murf > > > > > >> >>> -- >>> _ >>> -- 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] How to test BIG traffic through DAHDI/WANPIPEinterfaces
Hello my friend Ingmar, I would like to know the cable you used? how was the connection? i'm using this one: http://wiki.sangoma.com/Pinouts#A108 Loop Back Is this ok? what should i do my friend, my problems are "understand" the fisicall connection :( Best Regards!!! 2010/9/24 Ingmar Steen > Hi DD, > > > > We usually use loopback cables and use the open source SIP test tool “SIPp” > to initiate SIP calls that are sent from one group of 4 ports to another > group of 4 ports. > > > > Met vriendelijke groet, > > Ingmar Steen > > Teleknowledge > > > > *Van:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias > *Verzonden:* vrijdag 24 september 2010 11:05 > *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion > *Onderwerp:* [asterisk-users] How to test BIG traffic through > DAHDI/WANPIPEinterfaces > > > > Hello Community, > > > > I need to test or simulate many calls through dahdi/wanpipe, i have a > Sangoma A108D, and i need to test the stability of the > card/drivers/firmwares with a test environment, do you think is possible? > > > > What should i do? using some loopback cable maybe? > > > > Thanks in advance > > > > DD > > -- > _ > -- 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] No translator path exists for channel type DAHDI (native 76) to 256
Hello,
We are having issues with a NEW Sangoma A108D:
-- Executing [691918...@pbx1:1] Dial("SIP/xtravoip200-009d24b0",
"DAHDI/g0/691918892|30|m") in new stack
[Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No translator
path exists for channel type DAHDI (native 76) to 256
[Oct 1 10:04:43] WARNING[14171]: app_dial.c:1237 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 58 - Bearer capability not available)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [691918...@pbx1:2] Hangup("SIP/xtravoip200-009d24b0", "")
in new stack
== Spawn extension (pbx1, 691918892, 2) exited non-zero on
'SIP/xtravoip200-009d24b0'
wanrouter status:
wanrouter status
Devices currently active:
wanpipe1 wanpipe2 wanpipe3 wanpipe4 wanpipe5 wanpipe6 wanpipe7 wanpipe8
Wanpipe Config:
Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud
rate |
wanpipe1| N/A | A101/1D/A102/2D/4/4D/8| 17 | 4 | 1|
N/A | 0 |
wanpipe2| N/A | A101/1D/A102/2D/4/4D/8| 17 | 4 | 1|
N/A | 0 |
wanpipe3| N/A | A101/1D/A102/2D/4/4D/8| 17 | 4 | 1|
N/A | 0 |
wanpipe4| N/A | A101/1D/A102/2D/4/4D/8| 17 | 4 | 1|
N/A | 0 |
wanpipe5| N/A | A101/1D/A102/2D/4/4D/8| 17 | 4 | 1|
N/A | 0 |
wanpipe6| N/A | A101/1D/A102/2D/4/4D/8| 17 | 4 | 1|
N/A | 0 |
wanpipe7| N/A | A101/1D/A102/2D/4/4D/8| 17 | 4 | 1|
N/A | 0 |
wanpipe8| N/A | A101/1D/A102/2D/4/4D/8| 17 | 4 | 1|
N/A | 0 |
Wanrouter Status:
Device name | Protocol | Station | Status|
wanpipe1| AFT TE1 | N/A | Connected |
wanpipe2| AFT TE1 | N/A | Disconnected |
wanpipe3| AFT TE1 | N/A | Disconnected |
wanpipe4| AFT TE1 | N/A | Disconnected |
wanpipe5| AFT TE1 | N/A | Disconnected |
wanpipe6| AFT TE1 | N/A | Disconnected |
wanpipe7| AFT TE1 | N/A | Disconnected |
wanpipe8| AFT TE1 | N/A | Disconnected |
r...@sangoma-testing:/etc/asterisk# wanpipemon -i w1g1 -c Ta
* w1g1: E1 Alarms (Framer) *
ALOS:OFF| LOS:OFF
RED:OFF| AIS:OFF
OOF:OFF| RAI:OFF
* w1g1: E1 Alarms (LIU) *
Short Circuit:OFF
Open Circuit:OFF
Loss of Signal:OFF
* w1g1: E1 Performance Monitoring Counters *
Line Code Violation: 187
Far End Block Errors: 0
CRC4 Errors: 0
FAS Errors: 0
Rx Level: > -2.5db
r...@sangoma-testing:/etc/asterisk# asterisk -rx 'pri show spans'
PRI span 1/0: Provisioned, Up, Active
PRI span 2/0: Provisioned, In Alarm, Down, Active
PRI span 3/0: Provisioned, In Alarm, Down, Active
PRI span 4/0: Provisioned, In Alarm, Down, Active
PRI span 5/0: Provisioned, In Alarm, Down, Active
PRI span 6/0: Provisioned, In Alarm, Down, Active
PRI span 7/0: Provisioned, In Alarm, Down, Active
PRI span 8/0: Provisioned, In Alarm, Down, Active
What's happening?
Hope you can help me
Regards!
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Re: [asterisk-users] Kernel Panic When restarting the server
Thanks Tim That solved my problem, thank you very much...but now i'm having another problem, when the server starts, it doesn't start asterisk automatically, should i change the start script? 2010/9/30 Tim Nelson > - "Danny Dias" wrote: > >I'm getting a KErnel Pannic every time i restart the server, what could be > happening? > >I just make: "shutdown -r now" and the server gets Kernel Panic. I'have to > go on site and press the power button > > > I'd be willing to bet Wanpipe is attempting to stop while Asterisk is still > using it. It's a known problem. Put the following into > /etc/wanpipe/scripts/stop > > #!/bin/sh > /etc/init.d/asterisk stop > sleep 2 > /etc/init.d/asterisk stop > > Then chmod +x it. When Wanpipe attempts to stop, it will shut down Asterisk > first, bypassing a panic situation. > > --Tim > > -- > _ > -- 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] Kernel Panic When restarting the server
Hello, I'm getting a KErnel Pannic every time i restart the server, what could be happening? I just make: "shutdown -r now" and the server gets Kernel Panic. I'have to go on site and press the power button Here you have my sotware versions: Asterisk 1.4.24.1 DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 libpri version: 1.4.10.1 WANPIPE Release: 3.5.4 IS there something that i shoud check? Best Regards! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weird Behavior with DAHDI
Hello,
I'm experiencing some weird problems on my server:
- 1) The following messages are filling up my logs:
[Sep 29 08:24:59] WARNING[7077]: chan_dahdi.c:2789 pri_find_dchan: No
D-channels available! Using Primary channel 140 as D-channel anyway!
[Sep 29 08:24:59] WARNING[7078]: chan_dahdi.c:2789 pri_find_dchan: No
D-channels available! Using Primary channel 171 as D-channel anyway!
[Sep 29 08:24:59] WARNING[7075]: chan_dahdi.c:2789 pri_find_dchan: No
D-channels available! Using Primary channel 78 as D-channel anyway!
[Sep 29 08:24:59] WARNING[7079]: chan_dahdi.c:2789 pri_find_dchan: No
D-channels available! Using Primary channel 202 as D-channel anyway!
[Sep 29 08:24:59] WARNING[7073]: chan_dahdi.c:2789 pri_find_dchan: No
D-channels available! Using Primary channel 16 as D-channel anyway!
[Sep 29 08:24:59] WARNING[7080]: chan_dahdi.c:2789 pri_find_dchan: No
D-channels available! Using Primary channel 233 as D-channel anyway!
[Sep 29 08:24:59] WARNING[7074]: chan_dahdi.c:2789 pri_find_dchan: No
D-channels available! Using Primary channel 47 as D-channel anyway!
In the Asterisk CLI>, i'm watching these messages constantly
2) I've plugged in a real E1 PRI ISDN:
r...@sangoma-testing:/usr/src# asterisk -rx 'pri show spans'
PRI span 1/0: Provisioned, In Alarm, Down, Active
PRI span 2/0: Provisioned, In Alarm, Down, Active
PRI span 3/0: Provisioned, In Alarm, Down, Active
*PRI span 4/0: Provisioned, Up, Active*
PRI span 5/0: Provisioned, In Alarm, Down, Active
PRI span 6/0: Provisioned, In Alarm, Down, Active
PRI span 7/0: Provisioned, In Alarm, Down, Active
PRI span 8/0: Provisioned, In Alarm, Down, Active
Seems to be OK! but i can't make a call:
-- Executing [691918...@pbx1:1] Dial("SIP/xtravoip200-021a47e0",
"DAHDI/g4/691918892|30|m") in new stack
[Sep 29 08:29:51] WARNING[7338]: channel.c:3170 ast_request: No translator
path exists for channel type DAHDI (native 76) to 256
[Sep 29 08:29:51] WARNING[7338]: app_dial.c:1237 dial_exec_full: *Unable to
create channel of type 'DAHDI'* (cause 58 - Bearer capability not available)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [691918...@pbx1:2] Hangup("SIP/xtravoip200-021a47e0", "")
in new stack
== Spawn extension (pbx1, 691918892, 2) exited non-zero on
'SIP/xtravoip200-021a47e0'
What is happening?
Could you let me know how to debug or to understand the output from "pri
intense debug span 4"?
> TEI: 0 State 7(Multi-frame established)
> V(A)=30, V(S)=30, V(R)=30
> K=7, RC=0, l3initiated=0, reject_except=0, ack_pend=0
> T200_id=0, N200=3, T203_id=0
> [ 00 01 01 3d ]
> Supervisory frame:
> SAPI: 00 C/R: 0 EA: 0
> TEI: 000EA: 1
> Zero: 0 S: 0 01: 1 [ RR (receive ready) ]
> N(R): 030 P/F: 1
> 0 bytes of data
-- Starting T200 timer
Sangoma-Testing*CLI>
< TEI: 0 State 8(Timer recovery)
< V(A)=30, V(S)=30, V(R)=30
< K=7, RC=0, l3initiated=0, reject_except=0, ack_pend=0
< T200_id=1, N200=3, T203_id=0
< [ 02 01 01 3d ]
< Supervisory frame:
< SAPI: 00 C/R: 1 EA: 0
< TEI: 000EA: 1
< Zero: 0 S: 0 01: 1 [ RR (receive ready) ]
< N(R): 030 P/F: 1
< 0 bytes of data
> TEI: 0 State 8(Timer recovery)
> V(A)=30, V(S)=30, V(R)=30
> K=7, RC=0, l3initiated=0, reject_except=0, ack_pend=0
> T200_id=1, N200=3, T203_id=0
> [ 02 01 01 3d ]
Supervisory frame:>
> SAPI: 00 C/R: 1 EA: 0
> TEI: 000EA: 1
> Zero: 0 S: 0 01: 1 [ RR (receive ready) ]
> N(R): 030 P/F: 1
> 0 bytes of data
-- Got ACK for N(S)=30 to (but not including) N(S)=30
Done handling message for SAPI/TEI=0/0
Sangoma-Testing*CLI>
< TEI: 0 State 8(Timer recovery)
< V(A)=30, V(S)=30, V(R)=30
< K=7, RC=0, l3initiated=0, reject_except=0, ack_pend=0
< T200_id=1, N200=3, T203_id=0
< [ 00 01 01 3d ]
< Supervisory frame:
< SAPI: 00 C/R: 0 EA: 0
< TEI: 000EA: 1
< Zero: 0 S: 0 01: 1 [ RR (receive ready) ]
< N(R): 030 P/F: 1
< 0 bytes of data
-- Got ACK for N(S)=30 to (but not including) N(S)=30
-- Stopping T200 timer
-- Starting T203 timer
What should i check on the above span debug? what's important there? the
timers? the ACK? SAPI? TEI? is there any place to learn how to understand
this output?
Hope you can help me
Verions of my server:
libpri version: 1.4.11.4
Asterisk 1.4.24.1
DAHDI Version: 2.4.0
WANPIPE Release: 3.5.15.4
Please don't ask me to upgrade my Asterisk Version, the idea is to test this
environment
Best Regards!
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Re: [asterisk-users] NAT issue (i think?)
Hello Ron.. The answer that i see here is only a trying to a Register...means the REGISTRATION procedures are taking a significant amount of time. You should get a 200 OK Can you lease make a simple draw of your architecture? seems to be a NAT problem, that's for sure REgards! 2010/9/28 Ron > Hi Danny > > On the pap2 by default it is set to 3600 and i have not change that. > by the way, is the NAT keep-alive same with the NOTIFY message? coz i am > seeing my asterisk respond to those as bad event could that be causing > it to loose the registration? > > here's the registration from ngrep: > > U 78.65.34.12:5094 -> 12.34.56.78:5060 > REGISTER sip:sip.mydomain.com SIP/2.0. > Via: SIP/2.0/UDP 78.65.34.12:5094;branch=z9hG4bK-25a5eec4;rport. > From: Kristine > >;tag=68fc368d164925e0o0. > To: Kristine > >. > Call-ID: c9bd8b57-f7bdc...@192.168.1.52. > CSeq: 116228 REGISTER. > Max-Forwards: 70. > Contact: Kristine ;expires=3600. > User-Agent: Linksys/PAP2T-3.1.15(LS). > Content-Length: 0. > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. > Supported: x-sipura. > . > > > U 12.34.56.78:5060 -> 78.65.34.12:5094 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP > 78.65.34.12:5094;branch=z9hG4bK-25a5eec4;received=78.65.34.12;rport=5094. > From: Kristine > >;tag=68fc368d164925e0o0. > To: Kristine > >. > Call-ID: c9bd8b57-f7bdc...@192.168.1.52. > CSeq: 116228 REGISTER. > User-Agent: Asterisk PBX. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. > Supported: replaces. > Content-Length: 0. > > > On 9/28/10 7:24 PM, Danny Dias wrote: > > You have to increase the time of expiration for the Register...on linksys > > devices is located on Proxy and Registration section under the EXTN: > (Where > > N is the extension number) > > > > Try putting this to: 3600 > > > > To check wheter or not is loosing Register, try with ngrep-sip and check > it: > > > > ngrep -p -q -W byline port 5060>register.pkt > > > > Then post here the content of register.pkt; but please, after issuing the > > change explained above! > > > > Regards! > > > > 2010/9/28 Ron > > > >> Hi All. > >> > >> got this problem that IP phones could not re-register to my server. even > >> if device is power cycled it still would not register. the solution i > >> found was to change the SIP port settings on the phone and it will > >> register. but after registration expires and its time to re-register the > >> same thing will happen, so i have to update the port settings again just > >> to make it work which is troublesome. > >> > >> i'm using Asterisk 1.4.31 with the following realtime config: > >> > >> rtcachefriends=yes > >> rtsavesysname=yes > >> rtupdate=yes > >> rtautoclear=no > >> > >> one thing i noticed is that it only seems to happen on linksys devices > >> e.g. PAP2 and SPA's. another phone i'm using is yealink and so far no > >> client has complain about it. > >> > >> hope anyone can help. thank you. > >> > >> regards > >> Ron > >> > >> > >> -- > >> _ > >> -- 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] What's the meaning of this?
Hello, I'm checking this: [Sep 28 13:32:46] NOTICE[30360] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Sep 28 13:32:46] NOTICE[30363] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 [Sep 28 13:32:46] NOTICE[30363] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 [Sep 28 13:32:46] NOTICE[30361] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 What should i do? the calls are going down. And the Telco says that the E1's are ok -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT issue (i think?)
You have to increase the time of expiration for the Register...on linksys devices is located on Proxy and Registration section under the EXTN: (Where N is the extension number) Try putting this to: 3600 To check wheter or not is loosing Register, try with ngrep-sip and check it: ngrep -p -q -W byline port 5060 >register.pkt Then post here the content of register.pkt; but please, after issuing the change explained above! Regards! 2010/9/28 Ron > Hi All. > > got this problem that IP phones could not re-register to my server. even > if device is power cycled it still would not register. the solution i > found was to change the SIP port settings on the phone and it will > register. but after registration expires and its time to re-register the > same thing will happen, so i have to update the port settings again just > to make it work which is troublesome. > > i'm using Asterisk 1.4.31 with the following realtime config: > > rtcachefriends=yes > rtsavesysname=yes > rtupdate=yes > rtautoclear=no > > one thing i noticed is that it only seems to happen on linksys devices > e.g. PAP2 and SPA's. another phone i'm using is yealink and so far no > client has complain about it. > > hope anyone can help. thank you. > > regards > Ron > > > -- > _ > -- 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] Problems compiling Asterisk on Debian
r...@sangoma-testing:/home# cat /lib/modules/2.6.26-2-amd64/build/.config cat: /lib/modules/2.6.26-2-amd64/build/.config: No such file or directory r...@sangoma-testing:/home# cat /usr/src/linux/.config cat: /usr/src/linux/.config: No such file or directory r...@sangoma-testing:/home# uname -a Linux Sangoma-Testing 2.6.26-2-amd64 #1 SMP Thu Sep 16 15:56:38 UTC 2010 x86_64 GNU/Linux r...@sangoma-testing:/home# uname -r 2.6.26-2-amd64 2010/9/28 Paul Belanger > On Mon, Sep 27, 2010 at 6:57 PM, Danny Dias > wrote: > > r...@sangoma-testing:/home# ls -la /lib/modules/ > > total 12 > > drwxr-xr-x 3 root root 4096 2010-09-24 10:21 . > > drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. > > drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 > > > $ cat /lib/modules/2.6.26-2-amd64/build/.config > > > r...@sangoma-testing:/home# ls -la /usr/src/linux > > lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux -> > > > $ cat /usr/src/linux/.config > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > blog.polybeacon.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems compiling Asterisk on Debian
Sorry Paul, My mistake...i repeat the commands again: r...@sangoma-testing:/usr/src# ls -la /lib/modules/ total 28 drwxr-xr-x 7 root root 4096 2010-09-27 19:35 . drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. drwxr-xr-x 3 root root 4096 2010-09-27 19:29 2.6.26-1-amd64 drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 drwxr-xr-x 2 root root 4096 2010-09-27 19:35 2.6.26-2-openvz-amd64 drwxr-xr-x 2 root root 4096 2010-09-27 19:35 2.6.26-2-vserver-amd64 drwxr-xr-x 2 root root 4096 2010-09-27 19:35 2.6.26-2-xen-amd64 r...@sangoma-testing:/usr/src# ls -la /usr/src/linux ls: cannot access /usr/src/linux: No such file or directory Is that Ok? 2010/9/28 Danny Dias > Hello Paul, > > Here is the output of the commands: > > r...@sangoma-testing:/home# ls -la /lib/modules/ > total 12 > drwxr-xr-x 3 root root 4096 2010-09-24 10:21 . > drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. > drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 > > r...@sangoma-testing:/home# ls -la /usr/src/linux > lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux -> > linux-headers-2.6.26-2-amd64 > > Seems to be OK, isn't? > > Thanks! > > > 2010/9/27 Paul Belanger > >> On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias >> wrote: >> >> > The same problem! >> > >> What is the output from the following? >> >> $ ls -la /lib/modules/ >> >> $ ls -la /usr/src/linux >> >> -- >> Paul Belanger | dCAP >> Polybeacon | Consultant >> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) >> blog.polybeacon.com >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems compiling Asterisk on Debian
Thanks Jim, What do you mean with "redo" ? I did not run the ./configure, i'm installing dahdi-linux and just need : make && make install The problem is when i issue make Thanks for your answer my friend! 2010/9/28 Jim Dickenson > Did you install the header files after ./configure was run? If so redo the > ./configure command and see what that does. > -- > Jim Dickenson > mailto:dicken...@cfmc.com > > CfMC > http://www.cfmc.com/ > > > > On Sep 27, 2010, at 3:57 PM, Danny Dias wrote: > > Hello Paul, > > Here is the output of the commands: > > r...@sangoma-testing:/home# ls -la /lib/modules/ > total 12 > drwxr-xr-x 3 root root 4096 2010-09-24 10:21 . > drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. > drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 > > r...@sangoma-testing:/home# ls -la /usr/src/linux > lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux -> > linux-headers-2.6.26-2-amd64 > > Seems to be OK, isn't? > > Thanks! > > > 2010/9/27 Paul Belanger > >> On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias >> wrote: >> > The same problem! >> > >> What is the output from the following? >> >> $ ls -la /lib/modules/ >> >> $ ls -la /usr/src/linux >> >> -- >> Paul Belanger | dCAP >> Polybeacon | Consultant >> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) >> blog.polybeacon.com >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _ > -- 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] Problems compiling Asterisk on Debian
Hello Paul, Here is the output of the commands: r...@sangoma-testing:/home# ls -la /lib/modules/ total 12 drwxr-xr-x 3 root root 4096 2010-09-24 10:21 . drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 r...@sangoma-testing:/home# ls -la /usr/src/linux lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux -> linux-headers-2.6.26-2-amd64 Seems to be OK, isn't? Thanks! 2010/9/27 Paul Belanger > On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias > wrote: > > The same problem! > > > What is the output from the following? > > $ ls -la /lib/modules/ > > $ ls -la /usr/src/linux > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > blog.polybeacon.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems compiling Asterisk on Debian
The same problem! r...@sangoma-testing:/usr/src/dahdi-linux-complete-2.4.0+2.4.0# make all && make install && make config make -C linux all make[1]: Entering directory `/usr/src/dahdi-linux-2.1.0.4/dahdi-linux-complete-2.4.0+2.4.0/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/dahdi-linux-2.1.0.4/dahdi-linux-complete-2.4.0+2.4.0/linux/drivers/dahdi/firmware' Attempting to download dahdi-fwload-vpmadt032-1.25.0.tar.gz --2010-09-27 13:07:26-- http://downloads.digium.com/pub/telephony/firmware/releases/dahdi-fwload-vpmadt032-1.25.0.tar.gz Resolving downloads.digium.com... 76.164.171.232, 2001:470:e0d4::e8 Connecting to downloads.digium.com|76.164.171.232|:80... connected. HTTP request sent, awaiting response... 200 OK Length: 149360 (146K) [application/x-gzip] Saving to: `dahdi-fwload-vpmadt032-1.25.0.tar.gz' 100%[==>] 149,360 81.0K/s in 1.8s 2010-09-27 13:07:28 (81.0 KB/s) - `dahdi-fwload-vpmadt032-1.25.0.tar.gz' saved [149360/149360] make[2]: Leaving directory `/usr/src/dahdi-linux-2.1.0.4/dahdi-linux-complete-2.4.0+2.4.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/dahdi-linux-2.1.0.4/dahdi-linux-complete-2.4.0+2.4.0/linux' make: *** [all] Error 2 2010/9/27 Paul Belanger > On Mon, Sep 27, 2010 at 12:09 PM, Danny Dias > wrote: > > What should i do? > > > Try with the lastest DAHDI version, 2.4.0. > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > blog.polybeacon.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems compiling Asterisk on Debian
I've these versions of DAHDI running into another Server with no problem...it seems to be a problem with dependencies, but i can't find the trick :( 2010/9/27 Paul Belanger > On Mon, Sep 27, 2010 at 12:09 PM, Danny Dias > wrote: > > What should i do? > > > Try with the lastest DAHDI version, 2.4.0. > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > blog.polybeacon.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems compiling Asterisk on Debian
¿ahhh? 2010/9/27 Roger Burton West > On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote: > > >What should i do? > > aptitude install module-assistant > m-a a-i dahdi > > > -- > _ > -- 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] Problems compiling Asterisk on Debian
Done my friend: r...@sangoma-testing:/lib/modules/2.6.26-2-amd64/kernel/drivers# module-assistant prepare Getting source for kernel version: 2.6.26-2-amd64 apt-get install linux-headers-2.6.26-2-amd64 Reading package lists... Done Building dependency tree Reading state information... Done linux-headers-2.6.26-2-amd64 is already the newest version. 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded. Creating symlink... apt-get install build-essential Reading package lists... Done Building dependency tree Reading state information... Done build-essential is already the newest version. 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded. Done! r...@sangoma-testing:/lib/modules/2.6.26-2-amd64/kernel/drivers# cd /usr/src/dahdi-linux-2.1.0.4/ r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed." You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. exit 1 make: *** [modules] Error 1 The same result :( 2010/9/27 Daniel Tryba > On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote: > > r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make > > echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel > > installed." > > You do not appear to have the sources for the 2.6.26-2-amd64 kernel > > installed. > > exit 1 > > make: *** [modules] Error 1 > > > > What should i do? > > The easiest way would be to use module-assistant > # aptitude install module-assistant > # module-assistant prepare > > This checks which kernel you are running and install the right packages, > eg: > module-assistant prepare > Getting source for kernel version: 2.6.26-2-vserver-amd64 > apt-get install linux-headers-2.6.26-2-vserver-amd64 > Reading package lists... Done > Building dependency tree > Reading state information... Done > The following extra packages will be installed: > linux-headers-2.6.26-2-common-vserver linux-kbuild-2.6.26 > The following NEW packages will be installed: > linux-headers-2.6.26-2-common-vserver > linux-headers-2.6.26-2-vserver-amd64 > linux-kbuild-2.6.26 > 0 upgraded, 3 newly installed, 0 to remove and 0 not upgraded. > Need to get 4366kB of archives. > After this operation, 35.8MB of additional disk space will be > used. > Do you want to continue [Y/n]? > > -- > > Daniel Tryba > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- 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] Problems compiling Asterisk on Debian
Thanks Dean, I've done it before, that's why i'm here asking :( take a look: r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# apt-cache search linux-headers-$(uname -r) linux-headers-2.6.26-2-amd64 - Header files for Linux 2.6.26-2-amd64 r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# sudo apt-get install linux-headers-$(uname -r) Reading package lists... Done Building dependency tree Reading state information... Done linux-headers-2.6.26-2-amd64 is already the newest version. 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded. r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed." You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. exit 1 make: *** [modules] Error 1 2010/9/27 Dean Hoover > > Source files aren't automatically installed on every install. > > This link should help: > http://www.cyberciti.biz/faq/howto-install-kernel-headers-package/ > > Dean Hoover > Milwaukee, Wisconsin > > On 9/27/2010 11:09 AM, Danny Dias wrote: > > Hello, > > > > I'm trying to compile DAHDI on DEBIAN but i have the following error: > > > > r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make > > echo "You do not appear to have the sources for the 2.6.26-2-amd64 > > kernel installed." > > You do not appear to have the sources for the 2.6.26-2-amd64 kernel > > installed. > > exit 1 > > make: *** [modules] Error 1 > > > > What should i do? > > > > Thanks! > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems compiling Asterisk on Debian
Hello, I'm trying to compile DAHDI on DEBIAN but i have the following error: r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed." You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. exit 1 make: *** [modules] Error 1 What should i do? 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] How to test BIG traffic through DAHDI/WANPIPEinterfaces
Garet, MANY thanks my friend...can you believe that my brain was stucked :( So simple ;) THANKS for your valuable help! DD 2010/9/24 Gareth Blades > As the previous poster said use the sip software to make test calls. > Have the number it dials go out of the sangoma card and back into > another port via a crossover cable to an extension which answers and > plays back a file for a second or so before hanging up. > > You can then make lots of calls which constantly make outgoing calls on > 4 ports and incoming calls on another 4 ports. By being able to change > the diration of the call to can load the box very well. > > > Danny Dias wrote: > > ummm but how do you do that? > > SIPp is only for SIP calls...i need to check in some way the dahdi > > driver, i need in someway stress de card, is that possible? may be it > > has no sence at all :( > > > > Thanks! > > > > 2010/9/24 Ingmar Steen > <mailto:i.st...@teleknowledge.nl>> > > > > Hi DD, > > > > > > > > We usually use loopback cables and use the open source SIP test tool > > “SIPp” to initiate SIP calls that are sent from one group of 4 ports > > to another group of 4 ports. > > > > > > > > Met vriendelijke groet, > > > > Ingmar Steen > > > > Teleknowledge > > > > > > > > *Van:* asterisk-users-boun...@lists.digium.com > > <mailto:asterisk-users-boun...@lists.digium.com> > > [mailto:asterisk-users-boun...@lists.digium.com > > <mailto:asterisk-users-boun...@lists.digium.com>] *Namens *Danny > Dias > > *Verzonden:* vrijdag 24 september 2010 11:05 > > *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Onderwerp:* [asterisk-users] How to test BIG traffic through > > DAHDI/WANPIPEinterfaces > > > > > > > > Hello Community, > > > > > > > > I need to test or simulate many calls through dahdi/wanpipe, i have > > a Sangoma A108D, and i need to test the stability of the > > card/drivers/firmwares with a test environment, do you think is > > possible? > > > > > > > > What should i do? using some loopback cable maybe? > > > > > > > > Thanks in advance > > > > > > > > DD > > > > > > -- > > _ > > -- 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] How to test BIG traffic through DAHDI/WANPIPEinterfaces
ummm but how do you do that? SIPp is only for SIP calls...i need to check in some way the dahdi driver, i need in someway stress de card, is that possible? may be it has no sence at all :( Thanks! 2010/9/24 Ingmar Steen > Hi DD, > > > > We usually use loopback cables and use the open source SIP test tool “SIPp” > to initiate SIP calls that are sent from one group of 4 ports to another > group of 4 ports. > > > > Met vriendelijke groet, > > Ingmar Steen > > Teleknowledge > > > > *Van:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias > *Verzonden:* vrijdag 24 september 2010 11:05 > *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion > *Onderwerp:* [asterisk-users] How to test BIG traffic through > DAHDI/WANPIPEinterfaces > > > > Hello Community, > > > > I need to test or simulate many calls through dahdi/wanpipe, i have a > Sangoma A108D, and i need to test the stability of the > card/drivers/firmwares with a test environment, do you think is possible? > > > > What should i do? using some loopback cable maybe? > > > > Thanks in advance > > > > DD > > -- > _ > -- 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] How to test BIG traffic through DAHDI/WANPIPE interfaces
Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- 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] How to Understand a pri intense debug span X
Any hints please? I would appreciate your valuabl help Thanks 2010/9/16 Danny Dias > Hello my friends, > > I would like to understand the output from "pri intense debug span X", the > Telco always says that their side is OK, but i always receive alarms, > loosing connection, take a look: > > [Sep 16 13:18:19] WARNING[30364] chan_zap.c: Detected alarm on channel 1: > Recovering > [Sep 16 13:18:19] WARNING[30364] chan_zap.c: Unable to disable echo > cancellation on channel 1 > [...] > [Sep 16 13:18:24] NOTICE[30363] chan_zap.c: PRI got event: No more alarm > (5) on Primary D-channel of span 1 > > So, i'm working with pri intense debug span X, but the output is quite > difficult to understand, what should i check from this output? Where should > i find some information in order to make a debug and talk to my telco with > "power" :) > > Here is an example of pri intense debug span: > > < Supervisory frame: > [Sep 16 13:59:25] < SAPI: 00 C/R: 0 EA: 0 > < TEI: 000EA: 1 > [Sep 16 13:59:25] < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] > < N(R): 024 P/F: 1 > < 0 bytes of data > [Sep 16 13:59:25] Handling message for SAPI/TEI=0/0 > [Sep 16 13:59:25] -- ACKing all packets from 23 to (but not including) 24 > [Sep 16 13:59:25] -- Since there was nothing left, stopping T200 counter > [Sep 16 13:59:25] -- Stopping T203 counter since we got an ACK > [Sep 16 13:59:25] -- Nothing left, starting T203 counter > [Sep 16 13:59:25] -- Got RR response to our frame > [Sep 16 13:59:25] -- Restarting T203 counter > > Thanks in advance > > -- > Salu2 > > > -- Salu2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to Understand a pri intense debug span X
Hello my friends, I would like to understand the output from "pri intense debug span X", the Telco always says that their side is OK, but i always receive alarms, loosing connection, take a look: [Sep 16 13:18:19] WARNING[30364] chan_zap.c: Detected alarm on channel 1: Recovering [Sep 16 13:18:19] WARNING[30364] chan_zap.c: Unable to disable echo cancellation on channel 1 [...] [Sep 16 13:18:24] NOTICE[30363] chan_zap.c: PRI got event: No more alarm (5) on Primary D-channel of span 1 So, i'm working with pri intense debug span X, but the output is quite difficult to understand, what should i check from this output? Where should i find some information in order to make a debug and talk to my telco with "power" :) Here is an example of pri intense debug span: < Supervisory frame: [Sep 16 13:59:25] < SAPI: 00 C/R: 0 EA: 0 < TEI: 000EA: 1 [Sep 16 13:59:25] < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] < N(R): 024 P/F: 1 < 0 bytes of data [Sep 16 13:59:25] Handling message for SAPI/TEI=0/0 [Sep 16 13:59:25] -- ACKing all packets from 23 to (but not including) 24 [Sep 16 13:59:25] -- Since there was nothing left, stopping T200 counter [Sep 16 13:59:25] -- Stopping T203 counter since we got an ACK [Sep 16 13:59:25] -- Nothing left, starting T203 counter [Sep 16 13:59:25] -- Got RR response to our frame [Sep 16 13:59:25] -- Restarting T203 counter Thanks in advance -- Salu2 -- _ -- 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] Problems with audio
Thanks Sebastian, It's the same firmware version for all our linksys phones...and we have hundreds of pbx's runnning this firmwares versions without any problem 2010/9/15 Sebastian > Hi, > > On 09/15/2010 04:04 PM, Danny Dias wrote: > > Hello, > > > > I'm having some problems with a total SIP Asterisk scenario, some > > extensions when make internal and outgoing calls can't hear very well > > the other party, not echo, not packet lostthe problem is that the > > volume seems to be very low...what could be happening? i'm not sure what > > to check > > > > I had this problem with an Asterisk setup few months ago. People outside > the company/setup would hear people on the Asterisk side very > faintly/low volume. Even after pushing the volume up on the phones to > max. In my case, upgrading the firmware of the Grandstream phones we > were using solved the problem. I don't know if this is your case as well > though. > > 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 > -- Salu2 -- _ -- 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] Problems with audio
Hello Adriá... We are using Linksys 942, softphones Xlite...it's a macro pbx, with almost 1000 users, we've checked the gain and volume on the phones :( 2010/9/15 Adrià Vidal > > > On Wed, Sep 15, 2010 at 6:08 PM, Danny Dias wrote: > >> Yes my friend...CONFIRMED!!! G729 on both sides >> >> > If the problem happen with SIP to SIP calls and with the same codec, the > problem is inside the phone. > > Check if you can pump up the volume inside his configuration. > > What phones are you using? > > -- > -- > Adrià Vidal > > > > -- > _ > -- 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 > -- Salu2 -- _ -- 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] Problems with audio
Yes my friend...CONFIRMED!!! G729 on both sides 2010/9/15 Ishfaq Malik > Have you checked that the codec order on the phone matched the order set > on the server? > > On Wed, 2010-09-15 at 17:04 +0200, Danny Dias wrote: > > Hello, > > > > > > I'm having some problems with a total SIP Asterisk scenario, some > > extensions when make internal and outgoing calls can't hear very well > > the other party, not echo, not packet lostthe problem is that the > > volume seems to be very low...what could be happening? i'm not sure > > what to check > > > > > > Thanks! > > > > -- > > Salu2 > > > > > > -- > > _ > > -- 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 > > -- > 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 > -- Salu2 -- _ -- 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] Problems with audio
Hello, I'm having some problems with a total SIP Asterisk scenario, some extensions when make internal and outgoing calls can't hear very well the other party, not echo, not packet lostthe problem is that the volume seems to be very low...what could be happening? i'm not sure what to check Thanks! -- Salu2 -- _ -- 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] How to avoid interruptions with DIGIUM
Thanks Miguel, Excellent TIP! :) I will try and let you know Best Regards! 2010/9/10 Miguel Molina > El 10/09/10 03:14, Danny Dias escribió: > > >There used to be a problem with some Dell servers though, but that > > was already fixed some weeks ago. > > > > HEllo Moises, > > > > How did you solve the problems with DELL servers? the problem is that > > my servers gets a Kernel Panic for no any reason...could you please > > tell me the solution with DELL servers please? > > > > Thanks! > I suggest you to upgrade the firmware of your sangoma card, looking at > the changelog: > > AFT A102dm Firmware Change Log > = > > > Release V37 > --- > Feb 22 2010 > Type: Recommended A102dm Firmware >Fixed PCIe dma timeouts. On some new Dell/HP servers the > PCIe timeouts >on dma transactions were causing PCI fatal error messages >in the logs. This has now been fixed. > > Grab the firmware for your card from here: > http://wiki.sangoma.com/sangoma-hardware#aft_firmware > > Regards, > > -- > Ing. Miguel Molina > Grupo de Tecnología > Millenium Phone Center > > > -- > _ > -- 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 > -- Salu2 -- _ -- 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] How to avoid interruptions with DIGIUM
>There used to be a problem with some Dell servers though, but that was already fixed some weeks ago. HEllo Moises, How did you solve the problems with DELL servers? the problem is that my servers gets a Kernel Panic for no any reason...could you please tell me the solution with DELL servers please? Thanks! 2010/9/10 Moises Silva > On Thu, Sep 9, 2010 at 11:08 AM, Danny Dias wrote: > >> Thanks Kevin, >> >> But today i saw a Kernel Panic into my server, for no any apparent >> reasondoes >> this parameter could help: pci=routeirq >> >> By the way, we are using DELL servers, i've also used Sangoma, and always >> the same problem >> >> Thanks! >> > > I'd like to know which problem you had with the Sangoma card as there are > no shared interrupt issues we know of. > > There used to be a problem with some Dell servers though, but that was > already fixed some weeks ago. > > Moises Silva > Senior Software Engineer > Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON > L3R 9R6 Canada > t. 1 905 474 1990 x128 | e. m...@sangoma.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 > -- Salu2 -- _ -- 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] How to avoid interruptions with DIGIUM
Thanks Kevin, But today i saw a Kernel Panic into my server, for no any apparent reasondoes this parameter could help: pci=routeirq By the way, we are using DELL servers, i've also used Sangoma, and always the same problem Thanks! 2010/9/9 Kevin P. Fleming > On 09/09/2010 09:01 AM, Tim Nelson wrote: > > - "Andrew Latham" wrote: > >> modprobe blacklisting may be of help... > >> > > > > The module sharing interrupts with the card is his storage controller > (megasas). Blacklisting the storage controller module? That is not a good > idea... > > No, but he may have other devices he isn't actually using that are > consuming interrupts, and disabling one or more of those may free up > interrupts to be utilized by the storage controller or the TDM interface > card. On some systems, though, the possible interrupt choices are > restricted per PCI/PCI Express slot, though, so yeah... moving the card > to another slot may also be required. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kflem...@digium.com > Check us out at www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Salu2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to avoid interruptions with DIGIUM
Hello Asterisk community, I'm experiencing some problems with a Digium TE4XXP, the thing is that i'm sharing IRQ with some megasas device: 169: 69917985 0 0 0 0 0 0 0 IO-APIC-level megasas, wct4xxp I've been searching here: http://ubuntuforums.org/showthread.php?t=254623 Should i try in the pass this parameter in the boot Kernel "*pci=routeirq*" just to check if this will help? Thanks! -- Salu2 -- _ -- 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] [SOLVED ]How to finish an AGI
I've done it ;)
This is what i did:
In the Macro:
[macro-check-call-limit-mercurios]
exten => s,1,Set(group_name=out_calls_user_${SIPCHANINFO(peername)})
exten => s,n,Set(GROUP()=${group_name})
exten => s,n,GotoIf($[${GROUP_COUNT(${group_name})} >
${MAX_OUT_CALLS_PER_USER}]?forbidden,1)
; EXITO:
exten => s,n,MacroExit
; FRACASO:
exten => forbidden,1,NoOp(*** llamada saliente bloqueada: el usuario
${SIPCHANINFO(peername)} tiene actualmente
${MATH(${GROUP_COUNT(${group_name})})-1,int)} llamadas salientes)
exten => forbidden,n,Set(toagi=1);en este caso la llamada no la cuelga la
macro como en endpoints la cuelga el agi.
exten => forbidden,n,Hangup()
Then, in the AGI:
AGI:
$AGI->exec(Macro,"check-call-limit-mercurios");
$limitada = $AGI->get_variable('toagi');
if ($limitada eq '1'){
$AGI->verbose("Superado el limite de llamadas salientes out.agi tira
el canal");
$AGI->hangup($chann);
}
#
#
By the way, is it necessary to Hangup the Macro if the AGI is already doing
this?
BR ;)
2010/9/3 Steve Edwards
> On Thu, 2 Sep 2010, Danny Dias wrote:
>
> Sorry for my poor explanation...what i'm trying to do is to invoke a Macro
>> from my AGI, like this:
>>
>> $agi->exec("Macro","check-call-limit");
>>
>> If the Macro checks that the group_name is bigger than a number specified
>> for every peer with setvar it should Hangup the call (frobidden,1 in the
>> Gotoif...) but this
>> is not happening, the AGI always continue with is process and it doesn´t
>> play attention to the Hangup in the macro, the macro is here:
>>
>> [macro-check-call-limit]
>> exten => s,1,Set(group_name=out_calls_user_${SIPCHANINFO(peername)})
>> exten => s,n,Set(GROUP()=${group_name})
>> exten => s,n,GotoIf($[${GROUP_COUNT(${group_name})} >
>> ${MAX_OUT_CALLS_PER_USER}] forbidden,1)
>> ; EXITO:
>> exten => s,n,MacroExit
>> ; FRACASO:
>> exten => forbidden,1,NoOp(*** llamada saliente bloqueada: el usuario
>> ${SIPCHANINFO(peername)} tiene actualmente
>> ${MATH(${GROUP_COUNT(${group_name})})-1,int)} llamadas
>> salientes)
>> exten => forbidden,n,Hangup(21) ; ISUP 21 = SIP 403 (Forbidden)
>>
>
> The concept of calling a macro from within an AGI seem convoluted, but may
> work. I've never tried it.
>
> Any particular reason you don't want to put the logic of the macro in your
> AGI?
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] How to finish an AGI
>Any particular reason you don't want to put the logic of the macro in your
AGI?
Yes...i've no idea how to do it...it's a PERL script, i'm already checking
how to do this...but it will be a little complicated :(
2010/9/3 Steve Edwards
> On Thu, 2 Sep 2010, Danny Dias wrote:
>
> Sorry for my poor explanation...what i'm trying to do is to invoke a Macro
>> from my AGI, like this:
>>
>> $agi->exec("Macro","check-call-limit");
>>
>> If the Macro checks that the group_name is bigger than a number specified
>> for every peer with setvar it should Hangup the call (frobidden,1 in the
>> Gotoif...) but this
>> is not happening, the AGI always continue with is process and it doesn´t
>> play attention to the Hangup in the macro, the macro is here:
>>
>> [macro-check-call-limit]
>> exten => s,1,Set(group_name=out_calls_user_${SIPCHANINFO(peername)})
>> exten => s,n,Set(GROUP()=${group_name})
>> exten => s,n,GotoIf($[${GROUP_COUNT(${group_name})} >
>> ${MAX_OUT_CALLS_PER_USER}] forbidden,1)
>> ; EXITO:
>> exten => s,n,MacroExit
>> ; FRACASO:
>> exten => forbidden,1,NoOp(*** llamada saliente bloqueada: el usuario
>> ${SIPCHANINFO(peername)} tiene actualmente
>> ${MATH(${GROUP_COUNT(${group_name})})-1,int)} llamadas
>> salientes)
>> exten => forbidden,n,Hangup(21) ; ISUP 21 = SIP 403 (Forbidden)
>>
>
> The concept of calling a macro from within an AGI seem convoluted, but may
> work. I've never tried it.
>
> Any particular reason you don't want to put the logic of the macro in your
> AGI?
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Salu2
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Re: [asterisk-users] How to finish an AGI
YES YES...that's what i want ;) so simple but i was so tired :( I will try it and let you know ;) THANKS my friend 2010/9/2 Danny Nicholas > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Dias > *Subject:* Re: [asterisk-users] How to finish an AGI > > > > >No nicolas...that's not what i want...by the way sound very complicated > :( > > > > >What i need is to FINISH THE AGI when the MACRO reachs the Hangup...dont > worry for the purpose of the macro, if the macro reachs the hangup the Agi > should stop working, but it continues with his job... :( > > Does the AGI have a SIG(HUP) = IGNORE (pardon the syntax since I don’t know > if it’s PERL/PHP/whatever)? If so, the AGI is “indestructible” (will finish > or have to be “killed”) You could have the macro set a variable at hangup > and kill the AGI when it returned > > AGI runs > > Macro runs > > Macro gets hangup > > Set xx=yes > > Returns to AGI > > If (xx=yes exit) > > > > -- > _ > -- 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 > -- Salu2 -- _ -- 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] How to finish an AGI
No nicolas...that's not what i want...by the way sound very complicated :( What i need is to FINISH THE AGI when the MACRO reachs the Hangup...dont worry for the purpose of the macro, if the macro reachs the hangup the Agi should stop working, but it continues with his job... :( 2010/9/2 Danny Nicholas > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Dias > *Subject:* Re: [asterisk-users] How to finish an AGI > > > > > > This isn’t really a task for AGI since it is by nature single-call > specific. As I interpret what I read, you are calling this AGI from within > a call and you want it to hang up all calls in a group when the group has > exceeded it’s group limit. If this is indeed the case, you should make a > cron job to poll asterisk and do a soft hangup on the group when call-limit > is exceeded. > > > > Steve (as usual) will have a better answer, but that’s my .02. > > -- > _ > -- 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 > -- Salu2 -- _ -- 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] How to finish an AGI
>What should i do to finish the macro if this macro reachs the Hangup?
I tried to say: "What should i do to finish the *AGI* if this macro reachs
the Hangup?"
2010/9/2 Danny Dias
> Hello Steven...
>
> Sorry for my poor explanation...what i'm trying to do is to invoke a Macro
> from my AGI, like this:
>
> $agi->exec("Macro","check-call-limit");
>
> If the Macro checks that the group_name is bigger than a number specified
> for every peer with setvar it should Hangup the call (frobidden,1 in the
> Gotoif...) but this is not happening, the AGI always continue with is
> process and it doesn´t play attention to the Hangup in the macro, the macro
> is here:
>
> [macro-check-call-limit]
> exten => s,1,Set(group_name=out_calls_user_${SIPCHANINFO(peername)})
> exten => s,n,Set(GROUP()=${group_name})
> exten => s,n,GotoIf($[${GROUP_COUNT(${group_name})} >
> ${MAX_OUT_CALLS_PER_USER}] forbidden,1)
> ; EXITO:
> exten => s,n,MacroExit
> ; FRACASO:
> exten => forbidden,1,NoOp(*** llamada saliente bloqueada: el usuario
> ${SIPCHANINFO(peername)} tiene actualmente
> ${MATH(${GROUP_COUNT(${group_name})})-1,int)} llamadas salientes)
> exten => forbidden,n,Hangup(21) ; ISUP 21 = SIP 403 (Forbidden)
>
>
> What should i do to finish the macro if this macro reachs the Hangup?
>
> Thanks for your help my friend!
>
>
> 2010/9/2 Steve Edwards
>
> On Thu, 2 Sep 2010, Danny Dias wrote:
>>
>> > I need to finish an AGI script when it invokes a macro from dialplan,
>> > how can i do that? it's quite confusing...the macro is making a hangup
>> > but the script continues
>>
>> I don't understand your question, but I'm guessing it has something to do
>> with:
>>
>> 1) How to continue an AGI if a hangup occurs during execution -- trap HUP.
>>
>> 2) How to execute an AGI after a hangup -- use deadagi() in the h
>> extension
>>
>> 3) The AGI is invoking a macro -- I have no clue with the level of detail
>> provided.
>>
>> --
>> Thanks in advance,
>> -
>> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
>> Newline Fax: +1-760-731-3000
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Salu2
>
>
--
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Re: [asterisk-users] How to finish an AGI
Hello Steven...
Sorry for my poor explanation...what i'm trying to do is to invoke a Macro
from my AGI, like this:
$agi->exec("Macro","check-call-limit");
If the Macro checks that the group_name is bigger than a number specified
for every peer with setvar it should Hangup the call (frobidden,1 in the
Gotoif...) but this is not happening, the AGI always continue with is
process and it doesn´t play attention to the Hangup in the macro, the macro
is here:
[macro-check-call-limit]
exten => s,1,Set(group_name=out_calls_user_${SIPCHANINFO(peername)})
exten => s,n,Set(GROUP()=${group_name})
exten => s,n,GotoIf($[${GROUP_COUNT(${group_name})} >
${MAX_OUT_CALLS_PER_USER}] forbidden,1)
; EXITO:
exten => s,n,MacroExit
; FRACASO:
exten => forbidden,1,NoOp(*** llamada saliente bloqueada: el usuario
${SIPCHANINFO(peername)} tiene actualmente
${MATH(${GROUP_COUNT(${group_name})})-1,int)} llamadas salientes)
exten => forbidden,n,Hangup(21) ; ISUP 21 = SIP 403 (Forbidden)
What should i do to finish the macro if this macro reachs the Hangup?
Thanks for your help my friend!
2010/9/2 Steve Edwards
> On Thu, 2 Sep 2010, Danny Dias wrote:
>
> > I need to finish an AGI script when it invokes a macro from dialplan,
> > how can i do that? it's quite confusing...the macro is making a hangup
> > but the script continues
>
> I don't understand your question, but I'm guessing it has something to do
> with:
>
> 1) How to continue an AGI if a hangup occurs during execution -- trap HUP.
>
> 2) How to execute an AGI after a hangup -- use deadagi() in the h
> extension
>
> 3) The AGI is invoking a macro -- I have no clue with the level of detail
> provided.
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
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[asterisk-users] How to finish an AGI
Hello community, I need to finish an AGI script when it invokes a macro from dialplan, how can i do that? it's quite confusing...the macro is making a hangup but the script continues Thanks -- Salu2 -- _ -- 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] Maximum Wait Time queue option
Take a look here: http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue <http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue>> Queue(queuename[|options][|URL][|announceoverride][|*timeout*][|AGI]) Hope it helps! 2010/8/30 Tino > Hello, > > Is there any option to set the maximum number of seconds a caller can wait > in a queue before being pulled out ? > > 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 > -- Saludos Danny Dias -- _ -- 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] BLF - Realtime & Asterisk
Hello Asterisk-Community, I'm having an error with my BLF configuration on my asterisk...i've configured the sip peer like this: [8250] type=friend callerid=Extensión 8250 <8250> canreinvite=no context=pbx9 dtmfmode=rfc2833 host=dynamic insecure=no language=es nat=yes pickupgroup= callgroup= qualify=2000 secret=cyx2mo type=friend username=8250 subscribecontext=pbx9 call-limit=100 disallow=all allow=g729 allow=g723 allow=ulaw allow=alaw allow=ilbc allow=gsm allow=h263 allow=h263p allow=h264 ***sip.conf*** [general] . . . . notifyringing=yes notifyhold=no rtupdate=yes rtcachefriends=yes ***extensions.conf*** [pbx9] exten => 8340,hint,SIP/8340 include => pruebas switch => Realtime/pbx9 at extensions In the Asterisk CLI i could see this message: [Jul 15 18:43:14] WARNING[15867]: chan_sip.c:15766 handle_request_subscribe: SUBSCRIBE failure: unrecognized format: 'multipart/related' pvt: subscribed: 0, stateid: -1, laststate: 0, dialogver: 0, subscribecont: 'pbx9', subscribeuri: '' asterisk -rx 'sip show subscriptions' Peer UserCall ID ExtensionLast state TypeMailbox 0 active SIP subscriptions asterisk -rx 'show hints' -= Registered Asterisk Dial Plan Hints =- 8340 at pbx9: SIP/8340 State:Idle Watchers 0 - 1 hints registered And phone does not show any light with the the extension 8349 in use... Thanks in advance for your help -- Saludos Danny Dias SkypeID: danny.dias1 -- _ -- 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] BLF with Realtime
Thanks as always Zeeshan ;) I've changed my configuration, take a look: [8250] type=friend callerid=Extensión 8250 <8250> canreinvite=no context=pbx9 dtmfmode=rfc2833 host=dynamic insecure=no language=es nat=yes pickupgroup= callgroup= qualify=2000 secret=cyx2mo type=friend username=8250 subscribecontext=pbx9 call-limit=100 disallow=all allow=g729 allow=g723 allow=ulaw allow=alaw allow=ilbc allow=gsm allow=h263 allow=h263p allow=h264 ***sip.conf*** [general] . . . . notifyringing=yes notifyhold=no rtupdate=yes rtcachefriends=yes ***extensions.conf*** [pbx9] exten => 8340,hint,SIP/8340 include => pruebas switch => Realtime/pbx9 at extensions In the Asterisk CLI i could see this message: [Jul 15 18:43:14] WARNING[15867]: chan_sip.c:15766 handle_request_subscribe: SUBSCRIBE failure: unrecognized format: 'multipart/related' pvt: subscribed: 0, stateid: -1, laststate: 0, dialogver: 0, subscribecont: 'pbx9', subscribeuri: '' asterisk -rx 'sip show subscriptions' Peer UserCall ID ExtensionLast state TypeMailbox 0 active SIP subscriptions asterisk -rx 'show hints' -= Registered Asterisk Dial Plan Hints =- 8340 at pbx9: SIP/8340 State:Idle Watchers 0 - 1 hints registered And phone does not show any light with the the extension 8349 in use... Thanks in advance for your help > Message: 4 > Date: Wed, 14 Jul 2010 10:24:39 -0400 > From: Zeeshan Zakaria > Subject: Re: [asterisk-users] BLF with Realtime > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: > > Content-Type: text/plain; charset="iso-8859-1" > > On asterisk 1.4 using real-time, subscribecontext field never worked for me > and I have to add the hints in extensions.conf. But once there, they work > just fine. > > Zeeshan A Zakaria > -- _ -- 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] WARNING[15867]: chan_sip.c:15766
Hello Asterisk-Community, I'm having an error with my BLF configuration on my asterisk...i've configured the sip peer like this: [8250] type=friend callerid=Extensión 8250 <8250> canreinvite=no context=pbx9 dtmfmode=rfc2833 host=dynamic insecure=no language=es nat=yes pickupgroup= callgroup= qualify=2000 secret=cyx2mo type=friend username=8250 subscribecontext=pbx9 call-limit=100 disallow=all allow=g729 allow=g723 allow=ulaw allow=alaw allow=ilbc allow=gsm allow=h263 allow=h263p allow=h264 ***sip.conf*** [general] . . . . notifyringing=yes notifyhold=no rtupdate=yes rtcachefriends=yes ***extensions.conf*** [pbx9] exten => 8340,hint,SIP/8340 include => pruebas switch => Realtime/p...@extensions In the Asterisk CLI i could see this message: [Jul 15 18:43:14] WARNING[15867]: chan_sip.c:15766 handle_request_subscribe: SUBSCRIBE failure: unrecognized format: 'multipart/related' pvt: subscribed: 0, stateid: -1, laststate: 0, dialogver: 0, subscribecont: 'pbx9', subscribeuri: '' asterisk -rx 'sip show subscriptions' Peer UserCall ID ExtensionLast state TypeMailbox 0 active SIP subscriptions asterisk -rx 'show hints' -= Registered Asterisk Dial Plan Hints =- 8...@pbx9: SIP/8340 State:Idle Watchers 0 - 1 hints registered And phone does not show any light with the the extension 8349 in use... Thanks in advance for your help -- Saludos Danny Dias SkypeID: danny.dias1 -- _ -- 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] BLF with Realtime
Hello Asterisk community, I'm trying to use BLF with Asterisk Realtime, i've been searching for some info but nothing seems to be clear, can anyone help me eith some ideas to make this work ok? I'va my dialplan with Realtime Thanks in advance -- Saludos Danny Dias SkypeID: danny.dias1 -- _ -- 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 on Ubuntu
Hello Asterisk users, I'm having a little problem with an Asterisk installation on Ubuntu, i had installed many asterisks on CentOS but never in Ubuntu, the problem is that Asterisk and DAHDI does not start at system start...i have to make "/etc/init.d/asterisk start" and "/etc/init.d/dahdi start" manually every time i reboot the machine (my laptop for testing) So, what should i do in order to solve this situation? Thanks in advance Regards -- Saludos Danny Dias SkypeID: danny.dias1 -- _ -- 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] IBM X3650 with Asterisk???
Hello Asterisk Community, Does somebody had used an IBM X3650 server with Asterisk? we would like to know if this servers are reliable and works OK with linux and Asterisk? Does some of you has this servier with asterisk on production? should we install an special BIOS (linux bios) in order to make this work better? Regards -- Saludos Danny Dias SkypeID: danny.dias1 -- _ -- 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] incoming ghost call
Hello asterisk users... We are having a little problem in our installation, we are using Asterisk 1.4.21.2 and zaptel 1.4.11 with a Digium TDM410P (3FXO + 1FXS), the problem is that when we disconnect the line from any of the fxo ports, we receive an incoming ghost call (using zap/x channel) it rings on the phone but we cant hear nothing...it's always doing the same everytime we disconnect the lines from the fxo We tried with a Sangoma card, and the problem went away, but we must use this digium card, we've tried with answer/hangup on polarityswitch with all the options, and we cant make this work ok, what should we do? Thanks in advance -- Saludos Danny Dias SkypeID: danny.dias1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to set up Fax on Asterisk - Using analog Fax machines and HT502 (or FXS of a Digium TDM410P)
Hello Asterisk users, We are having MANY but MANY problems configuring an analog fax machine to work properly on Asterisk, the first thing we do was to plug in the Fax analog machine to the FXS port of the Digium TDM410P, we set echocancel=no on zapata.conf and also faxdetect=yes on general section, but our Asterisk CRASH every time we try to send/receive fax! We are using Asterisk 1.4.21 and Zaptel 1.4.11; the card does not shows any interrupt in /proc/interrupts We also tried with a Sangoma B600 on this machine and the same result! Then we tried with the sangoma and digium card on another asterisk box, with Asterisk 1.4.30 and DAHDI 2.2.x and the fax was not reliable 100% but at least Asterisk vener went down We cant make any upgrade of Asterisk/Zaptel due to some rules of the customer, the do not want to use fax2email, they need to use the panasonic fax machine, this is driving me crazy! We also tried with a HT502 with passtrough fax mode and pcmu and pcma enabled and the same result, asterisk does down when trying to receive/send a fax What could solve our problem? what else should we try about configuration? just faxdetect=incoming and set echocancel=yes and that's all? Please your help, we really need to put this working Thanks in advance for all your help! -- Saludos Danny Dias SkypeID: danny.dias1 -- _ -- 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] Problems with Fax over TDM410P
What do you mean with problems on my configuration? This is a FXO port on zapata: >> signalling=fxs_ks >> group=0 >> channel => 1 Not a FXS...can you explain to me what were you trying to say? > Message: 4 > Date: Mon, 12 Apr 2010 13:14:49 -0400 > From: David Backeberg > Subject: Re: [asterisk-users] Problems with Fax over TDM410P > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: > > Content-Type: text/plain; charset=ISO-8859-1 > > On Sun, Apr 11, 2010 at 5:00 PM, Danny Dias > wrote: > >> This digium card has 3 FXO ports and 1 FXS port where we have a fax > >> machine > >> connected! > >> > >> The problem is that we can receive fax very good, but we can't make any > >> outbound fax call, in fact, our asterisk get freezed in this case! > >> ; TDM410P > >> signalling=fxs_ks > >> group=0 > >> channel => 1 > >> > >> Signalling=fxs_ks > >> group=0 > >> channel => 2 > >> > >> signalling=fxs_ks > >> group=0 > >> channel => 3 > >> > >> signalling=fxo_ks > >> group=1 > >> channel => 4 > >> > >> What should we do in order to make it work ok? we really need to put > this > > If you really have three FXO, and one FXS, there's part of your > problem. You have your zapata configured as three FXS and one FXO. I > would suspect that would be a good enough reason to crash your card or > whatever. > > > > -- _ -- 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 DIES with no trace. PLEASE
Hello Zeeshan/Asterisk-users We are having a little problem in our Asterisk pbx using our A102DE, just like Zeeshan told us about problems with zap, even if a zap channel is in use the Hookstat is always onhook, never changes to offhook If the line is in use or not, the behavior of the Hookstate is always onhook, is this a problem? what should we do? MyPbx*CLI> zap show channel 31 Channel: 31I> File Descriptor: 44 Span: 1 Extension: I> Dialing: no Context: mde-g0 Caller ID: 2432690033 Calling TON: 33 Caller ID name: Destroy: 0LI> InAlarm: 0 Signalling Type: ISDN PRI Radio: 0*CLI> Owner: Real: Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently ON PRI Flags: I> PRI Logical Span: Implicit Hookstate (FXS only): Onhook Thanks in advance! > > Message: 1 > > Date: Thu, 18 Mar 2010 11:20:38 -0400 > > From: Zeeshan Zakaria > > Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > Message-ID: > ><5ad99e891003180820l56882922gd84102d158918...@mail.gmail.com> > > Content-Type: text/plain; charset="iso-8859-1" > > > > Do you properly hang up the calls. Does 'zap show channel number>' > > shows that the channel is 'on hook' after its hang up? > > > > On 2010-03-18 10:06 AM, "Danny Dias" wrote: > > > > Thanks Zeeshan, > > > > SAngoma told me that the asterisk problem is unrelated to wanpipe > drivers, > > they told me to reinstall asterisk again > > > > But, i still having doubts about the problem :( > > > > Thanks in advance > > > > > > > -- _ -- 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] Problems with Fax over TDM410P
Hi asterisk-users I'm really having big problems with this configuration, has anyone attached a fax machine to a FXS port of a digium tdm410P card succesfully? What changes should i do on asterisk to make this work ok? I just want to use this fax machine as a fax and not to voice! Thanks! > Message: 9 > Date: Fri, 9 Apr 2010 19:22:05 -0430 > From: Danny Dias > Subject: [asterisk-users] Problems with Fax over TDM410P > To: asterisk-users@lists.digium.com > Message-ID: > > Content-Type: text/plain; charset="iso-8859-1" > > Hello my friends... > > We are having some problems with the fax in our asterisk server... > > We have: > > Asterisk 1.4.21.2 > Zaptel Version: 1.4.11 > libpri version: 1.4.5 > Digium Card TDM 410P > > This digium card has 3 FXO ports and 1 FXS port where we have a fax machine > connected! > > The problem is that we can receive fax very good, but we can't make any > outbound fax call, in fact, our asterisk get freezed in this case! > > take a look in our zapata: > > [channels] > language=es > ;context=default > rxwink=300 > usecallerid=yes > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > canpark=yes > cancallforward=yes > callreturn=yes > echocancel=yes > echocancelwhenbridged=yes > rxgain=0.0 > txgain=0.0 > immediate=no > busydetect=yes > immediate=no > ;busycount=4 > ;busypattern=500,500 > ;answeronpolarityswitch=yes > ;hanguponpolarityswitch=yes > > > ; TDM410P > context = mde-g1 > immediate=no > signalling=fxs_ks > group=0 > channel => 1 > > context = mde-g1 > immediate=yes > Signalling=fxs_ks > group=0 > channel => 2 > > context = mde-g1 > immediate=yes > signalling=fxs_ks > group=0 > channel => 3 > > context=inside > faxdetect=incoming > immediate=no > signalling=fxo_ks > group=1 > channel => 4 > > What should we do in order to make it work ok? we really need to put this > working, i've heard that asterisk does not work very well with fax, but at > least it should try to dend it, not to get frozen :S > > Thanks in advance for all your help! > > Regards > -- next part -- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100409/ec63bd44/attachment-0001.htm > > -- _ -- 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] Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCode
Thanks James, What i need is to make the fax machines connected to the audiocodes mediant 1000 be able to send and receive fax throught Asterisk (connected to a pri) I know it's not reliable, but it should work at leaste, what should i do on Asterisk and Mediant to make this work? Im quite confuse with all these fax issues :S Thanks in advance > > Message: 11 > Date: Fri, 9 Apr 2010 17:30:23 -0700 > From: James Lamanna > Subject: Re: [asterisk-users] Fax Over PRI connected to a Sangoma card >- Fax machines connected to Sip Mediant AudioCodes > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: > > Content-Type: text/plain; charset=ISO-8859-1 > > On Fri, Apr 9, 2010 at 5:17 PM, Danny Dias > wrote: > > Hello my friends, > > I want to make fax work in the following scenario: > > My versions are: > > > > Asterisk 1.4.21.2 > > > > WANPIPE Release: 3.4.7 > > Zaptel Version: 1.4.11 > > libpri version: 1.4.5 > > Digium Card TDM 410P > > > > The E1 pri is connected to our Sangoma A102DE, we also have a SIP Mediant > > Audiocodes 1000 where we have some fax machines connected to fxs ports, > what > > we need is to make fax machines through mediant send faxes to the pstn > > (through E1 PRI) and viceversa... > > > > What should we do to make this work properly? what parameters in zapata? > > mediant 1000? > > > > Thanks in advance for all your help! > > I've had fairly good success with faxing using Asterisk + Hylafax. > I haven't tried any of the built-in Asterisk faxing programs yet > because I designed this setup before the newest revisions, when > Asterisk + built-in faxing was not working well. > What I do is run Hylafax on the same machine as Asterisk, and then run > IAXModem to do the communication between the 2. There's a lot of > documentation online about how to set this up. > > -- James > > > > > > > > > > > -- > > _ > > -- 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] Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCodes
Hello my friends, I want to make fax work in the following scenario: My versions are: Asterisk 1.4.21.2 WANPIPE Release: 3.4.7 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P The E1 pri is connected to our Sangoma A102DE, we also have a SIP Mediant Audiocodes 1000 where we have some fax machines connected to fxs ports, what we need is to make fax machines through mediant send faxes to the pstn (through E1 PRI) and viceversa... What should we do to make this work properly? what parameters in zapata? mediant 1000? Thanks in advance for all your help! -- _ -- 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] Problems with Fax over TDM410P
Hello my friends... We are having some problems with the fax in our asterisk server... We have: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P This digium card has 3 FXO ports and 1 FXS port where we have a fax machine connected! The problem is that we can receive fax very good, but we can't make any outbound fax call, in fact, our asterisk get freezed in this case! take a look in our zapata: [channels] language=es ;context=default rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no busydetect=yes immediate=no ;busycount=4 ;busypattern=500,500 ;answeronpolarityswitch=yes ;hanguponpolarityswitch=yes ; TDM410P context = mde-g1 immediate=no signalling=fxs_ks group=0 channel => 1 context = mde-g1 immediate=yes Signalling=fxs_ks group=0 channel => 2 context = mde-g1 immediate=yes signalling=fxs_ks group=0 channel => 3 context=inside faxdetect=incoming immediate=no signalling=fxo_ks group=1 channel => 4 What should we do in order to make it work ok? we really need to put this working, i've heard that asterisk does not work very well with fax, but at least it should try to dend it, not to get frozen :S Thanks in advance for all your help! Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Safe_asterisk doesn't exists???
> > Thanks Andrew, > > I have some doubts regarding this issue, firstly, i'm using Asterisk 1.4.21 > not 1.6 like the issue you showed to me ( > https://issues.asterisk.org/view.php?id=16887) other thing is that i have > many other asterisk servers working good and i never made this change > > By the way i'm using Centos and i can't find the line: > start-stop-daemon --start --oknodo --background --exec $DAEMON -- > $ASTARGS > > Is this a bug from Asterisk 1.6 only? > > Thanks in advance for your help my friend > > > >> -- >> >> Message: 4 >> Date: Wed, 24 Mar 2010 13:59:04 -0400 >> From: Andrew Latham >> Subject: Re: [asterisk-users] Safe_asterisk doesn't exists??? >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> >> Message-ID: >> >> Content-Type: text/plain; charset=UTF-8 >> >> https://issues.asterisk.org/view.php?id=16887 >> >> do a "make update" >> >> >> ~ >> Andrew "lathama" Latham >> lath...@gmail.com >> >> * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software >> * Learn more about Linux http://en.wikipedia.org/wiki/Linux >> * Learn more about Tux http://en.wikipedia.org/wiki/Tux >> >> >> >> On Tue, Mar 23, 2010 at 7:16 PM, Danny Dias >> wrote: >> > Hello my friends, >> > I'm very worry about a problem i'm having...my asterisk got freez some >> > times, every 5 or 6 days with NO trace in /var/log/asterisk/messages >> > What i want to know is if safe_asterisk has something to be with this? >> > This is what i have on my server: >> > [r...@mypbx ~]# ps -A | grep asterisk >> > ?9118 ? ? ? ? ?00:01:30 asterisk >> > [r...@dreampbx ~]# ps aux | grep asterisk >> > root ? ? ?9118 ?0.1 ?0.3 29668 12520 ? ? ? ? Sl ? Mar22 ? 1:30 >> > /usr/sbin/asterisk -f -vvvg -c >> > root ? ? 12096 ?0.0 ?0.0 ?4140 ?640 pts/1 ? ?S+ ? 18:40 ? 0:00 grep >> asterisk >> > I have another asterisk servers working and the commands above always >> shows >> > safe _asterisk as a process... >> > This safe_asterisk could be the cause of my problems? how does it works? >> how >> > can i activate it? >> > Thanks in advance for your valuable help! >> > DD >> > >> > -- >> > _ >> > -- 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] Safe_asterisk doesn't exists???
Thanks Andrew, I have some doubts regarding this issue, firstly, i'm using Asterisk 1.4.21 not 1.6 like the issue you showed to me ( https://issues.asterisk.org/view.php?id=16887) other thing is that i have many other asterisk servers working good and i never made this change By the way i'm using Centos and i can't find the line: start-stop-daemon --start --oknodo --background --exec $DAEMON -- $ASTARGS Is this a bug from Asterisk 1.6 only? Thanks in advance for your help my friend > -- > > Message: 4 > Date: Wed, 24 Mar 2010 13:59:04 -0400 > From: Andrew Latham > Subject: Re: [asterisk-users] Safe_asterisk doesn't exists??? > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: > > Content-Type: text/plain; charset=UTF-8 > > https://issues.asterisk.org/view.php?id=16887 > > do a "make update" > > > ~ > Andrew "lathama" Latham > lath...@gmail.com > > * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software > * Learn more about Linux http://en.wikipedia.org/wiki/Linux > * Learn more about Tux http://en.wikipedia.org/wiki/Tux > > > > On Tue, Mar 23, 2010 at 7:16 PM, Danny Dias > wrote: > > Hello my friends, > > I'm very worry about a problem i'm having...my asterisk got freez some > > times, every 5 or 6 days with NO trace in /var/log/asterisk/messages > > What i want to know is if safe_asterisk has something to be with this? > > This is what i have on my server: > > [r...@mypbx ~]# ps -A | grep asterisk > > ?9118 ? ? ? ? ?00:01:30 asterisk > > [r...@dreampbx ~]# ps aux | grep asterisk > > root ? ? ?9118 ?0.1 ?0.3 29668 12520 ? ? ? ? Sl ? Mar22 ? 1:30 > > /usr/sbin/asterisk -f -vvvg -c > > root ? ? 12096 ?0.0 ?0.0 ?4140 ?640 pts/1 ? ?S+ ? 18:40 ? 0:00 grep > asterisk > > I have another asterisk servers working and the commands above always > shows > > safe _asterisk as a process... > > This safe_asterisk could be the cause of my problems? how does it works? > how > > can i activate it? > > Thanks in advance for your valuable help! > > DD > > > > -- > > _ > > -- 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] Aastra weirds IP 169.x.x.x
Hello my friends... Currently we are using the following firmware versions on ours aastra 55i: Firmware Information Attribute Value Firmware Version 2.1.0.2145 Firmware Release Code SIP Boot Version 2.0.1.1055 Date/Time Jun 20 2007 06:20:29 Can we make a firmware upgrade to the latest one: 6755i (55i) SIP, V2.5.3.18, January 2010 , English , ZIP , 2,849 KB on the site: http://www.aastra.com/cps/rde/xchg/SID-3D8CCB6A-FE6670DF/04/hs.xsl/19705.htm Could this make problems? We are receving very weirds ip on the server, see here: mypbx*CLI> sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 10.4.1.130 (None) b7dd744679a 00101/14365 0x0 (nothing)No Rx: REGISTER 10.4.1.151 308 8cbe459c33e 00102/16076 0x0 (nothing)No Tx: NOTIFY 10.4.1.144 368 607d5af86cd 00102/25625 0x0 (nothing)No Tx: NOTIFY 169.254.236.26 308 4f407ce65eb 00102/15097 0x0 (nothing)No Tx: NOTIFY 169.254.21.164 309 f1e31e48e10 00102/23948 0x0 (nothing)No Tx: NOTIFY 5 active SIP channels Can you see the 169.x.x.x? whats the meaning of this? is this a serious firmware problem? dhcp problem? Thanks in advance DD -- _ -- 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] Safe_asterisk doesn't exists???
Thanks for all the answers... Asterisk starts at boot time, but if i "stop now" this is how i will make it up again: asterisk asterisk -vvvrc The weird think is that safe_Asterisk doesn't appear on my process, take a look: [r...@mypbx ~]# ps -A | grep asterisk 14605 ?00:00:02 asterisk 14704 pts/000:00:00 asterisk [r...@mypbx ~]# ps aux | grep asterisk root 14605 0.1 0.2 26396 10256 ? Sl 12:16 0:02 /usr/sbin/asterisk -f -vvvg -c root 14704 0.0 0.0 4216 1276 pts/0S+ 12:17 0:00 rasterisk r root 14819 0.0 0.0 4716 640 pts/3S+ 12:44 0:00 grep asterisk Sometimes my asterisk got frozen...i mean, stop now does not work, asterisk still running but nothing works, not even internall or outgoing calls, my only way out isto kill the process and start again... Thanks in advance for all your help! > > -- > > Message: 2 > Date: Wed, 24 Mar 2010 14:39:06 +0530 > From: Prince Singh > Subject: Re: [asterisk-users] Safe_asterisk doesn't exists??? > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: ><626964fc1003240209i6a48bd5cj93fbacd25e2cf...@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > @Danny: How do you start your Asterisk ? > > -- > Regards, > Prince Singh > > Drishti-Soft Solutions Pvt Ltd > > > On Wed, Mar 24, 2010 at 6:35 AM, Steve Edwards >wrote: > > > On Tue, 23 Mar 2010, Danny Dias wrote: > > > > > This safe_asterisk could be the cause of my problems? how does it > works? > > > how can i activate it? > > > > safe_asterisk is a script that usually lives at /usr/sbin/safe_asterisk. > > > > The script runs in the background. If it detects that Asterisk died, it > > can send you an email before restarting Asterisk. > > > > safe_asterisk is not the problem, but it can be useful as a band-aid > until > > you find the real problem. > > > > -- > > Thanks in advance, > > - > > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 > PST > > Newline Fax: +1-760-731-3000 > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- next part -- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100324/f2b441a4/attachment-0001.htm > > -- > > > -- _ -- 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] Safe_asterisk doesn't exists???
Hello my friends, I'm very worry about a problem i'm having...my asterisk got freez some times, every 5 or 6 days with NO trace in /var/log/asterisk/messages What i want to know is if safe_asterisk has something to be with this? This is what i have on my server: [r...@mypbx ~]# ps -A | grep asterisk 9118 ?00:01:30 asterisk [r...@dreampbx ~]# ps aux | grep asterisk root 9118 0.1 0.3 29668 12520 ? Sl Mar22 1:30 /usr/sbin/asterisk -f -vvvg -c root 12096 0.0 0.0 4140 640 pts/1S+ 18:40 0:00 grep asterisk I have another asterisk servers working and the commands above always shows safe _asterisk as a process... This safe_asterisk could be the cause of my problems? how does it works? how can i activate it? Thanks in advance for your valuable help! DD -- _ -- 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] How to make upgrades with Asterisk
Thanks Zeeshan, In fact,i have RealTime configured and working... What i want is to make an upgrade of libpri and wanpipe at least, asterisk and zaptel will be like i have now... Do you think that recompile/upgrade this softwares version will produce a problem? what steps should i do? Is it necessary to recompile asterisk if i make an upgrade of libpri? this recompilation will affect the realtime or the well bahavior of the server? Working with an old version of asterisk like 1.4.21.2 with the newest version of libpri is recommended or not? Thanks in advance for all your advices! > Message: 11 > Date: Mon, 22 Mar 2010 23:15:00 -0400 > From: Zeeshan Zakaria > Subject: Re: [asterisk-users] How to make upgrades with Asterisk > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: ><5ad99e891003222015o1727265cr4860d4e8a0b96...@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > If it is a production server, you should not do the upgrade on it. Setup a > new server with upgraded software, migrate all the data, test it and make > sure it works fine. There are things like CDR and voicemail which are > constantly being updated, meaning just before the final migration, you > should copy them to the new server. > > I have done some migrations and have found clonezilla to be a wonderful > tool > for this purpose. You create a second server on any computer, and once it > is > ready, clone it on a USB stick or CD, or on another computer via SSH, then > clone the production server for backup purposes on a medium of your choice, > and finally restore this new server image on to the production server. If > something goes wrong, you'll be able to restore the server back to its > functional state from the backup cloned image. > > When I migrated my own production server from 1.2 to 1.4, I did the > rehearsal many times, and very carefully drafted the whole migation plan. > This also included asking all the users to copy and delete their voicemails > before the day of migration. It took me about two weeks in planning and > making sure every single setting will be migrated, before I was comfotable > to do the migration, which took hardly an hour, and went just perfectly > smooth. > > Personally I am of the opinion that if it is not really necessary, don't > upgrade it. Will 1.6 give you something which you don't have in 1.4? It'll > have its own issues and learning curve. I tried it once and it was only a > pain for my setup, specially with real-time architecture, and a few other > things which I can't remember now. > > -- > Zeeshan A Zakaria > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to make upgrades with Asterisk
Hello my friends, I want to make upgrades for all my software, currently i have the following versions: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 WANPIPE Release: 3.4.7 libpri version: 1.4.5 I want to make upgrade for the last version of Asterisk 1.4, the last version of Zaptel (dahdi will be nice!), the newest libpri version and wanpipe What should i do? this is a production server and i don't want to mess something...is there a step by step to make this upgrade? Thanks in advance for your valuable help! DD -- _ -- 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
