Re: [asterisk-users] Cisco IP Phones and Asterisk
Good morning, from the last question i assume you're looking for a SIP-based configureation. On 03-30-2011 00:16, bilal ghayyad wrote: 1) How I can assign for each button an extension? you can configure them as lines (at least in my 7940). look for linex_name, linex_authname and linex_password settings in the config file. 2) How I can assign for specific button a feature to be used (like call forward or call pickup .. etc)? AFAIK you can't reprogram the softkeys. There are two buttons which you can use for programming (well, sort of). You can define the mailbox extension which can be any extension. You can write a dial plan for a specific function and then use it as voicemail. The other button is the service button which can be programmed to access any HTTP url. I'm using mine to switch my desk lamp on or off. 3) As you know that it is required to have a correct username and password to login, so where to give the username and password in the Cisco IP Phone to be able to login for the SIP account? same as 1) rgds a. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 and fail2ban
On Wed, 30 Mar 2011 01:45:20 +0300, Ioan Indreias indre...@gmail.com wrote: Just to provide an alternative to sshguard: you could use BFD[1] Thanks Ioan. I'll give it a shot. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 and fail2ban
Just to respond to the IP range approach. My ISP recently changed my external IP and now it appears that I am in New York (when I am actually static in Manchester, England). I've also been in Birmingham, Motherwell and Nottingham [UK] aswell! So, although banning certain ranges may be a good idea for you - it's not a good idea for everyone (we have 'road warriors' that do, indeed, travel to the Far East and Middle East). I suppose the only 'real' way to invoke security (on any system) is to have very strong passwords - maybe 1234 is not the way to go :p -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: 30 March 2011 10:08 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk and fail2ban On Wed, 30 Mar 2011 01:45:20 +0300, Ioan Indreias indre...@gmail.com wrote: Just to provide an alternative to sshguard: you could use BFD[1] Thanks Ioan. I'll give it a shot. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 and fail2ban
so does anyone use fail2ban w/ asterisk or most people use sshguard? On Wed, Mar 30, 2011 at 6:57 AM, Andrew Thomas a...@datavox.co.uk wrote: Just to respond to the IP range approach. My ISP recently changed my external IP and now it appears that I am in New York (when I am actually static in Manchester, England). I've also been in Birmingham, Motherwell and Nottingham [UK] aswell! So, although banning certain ranges may be a good idea for you - it's not a good idea for everyone (we have 'road warriors' that do, indeed, travel to the Far East and Middle East). I suppose the only 'real' way to invoke security (on any system) is to have very strong passwords - maybe 1234 is not the way to go :p -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: 30 March 2011 10:08 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk and fail2ban On Wed, 30 Mar 2011 01:45:20 +0300, Ioan Indreias indre...@gmail.com wrote: Just to provide an alternative to sshguard: you could use BFD[1] Thanks Ioan. I'll give it a shot. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 and fail2ban
I think you will find Fail2Ban the defacto standard. From: vip killa Sent: Wed 3/30/2011 8:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and fail2ban so does anyone use fail2ban w/ asterisk or most people use sshguard? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 and fail2ban
On Wed, Mar 30, 2011 at 9:38 AM, vip killa vipki...@gmail.com wrote: so does anyone use fail2ban w/ asterisk or most people use sshguard? Vip, the overall message is that it takes layers of settings/configurations to secure an installation. Simple Guide 1. alwaysauthreject = yes in http://svn.asterisk.org/svn/asterisk/trunk/configs/sip.conf.sample 2. Static firewall rules 2.1 Drop invalid traffic 2.2 Slow ICMP and TCP Reset attacks 2.3 Disable unneeded services 3. Dynamic firewall rules 3.1 Fail2ban (works ok, but you should test it) 3.2 Portscanning Block (http://www.newartisans.com/2007/09/neat-tricks-with-iptables.html) 3.3 Other solutions 3.4 Bad Network Lists (http://www.spamhaus.org/drop/) 4. Auditing. None of the above will work if not audited or reviewed on a regular basis. 5. Reporting. With Monthly reporting you can see trends and make good choices. -- ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 Packages for Debian and Ubuntu
On 11-03-29 07:59 AM, Daniel Pocock wrote: # apt-key adv --keyserver subkeys.pgp.net --recv-keys 175E41DF Executing: gpg --ignore-time-conflict --no-options --no-default-keyring --secret-keyring /etc/apt/secring.gpg --trustdb-name /etc/apt/trustdb.gpg --keyring /etc/apt/trusted.gpg --primary-keyring /etc/apt/trusted.gpg --keyserver subkeys.pgp.net --recv-keys 175E41DF gpg: requesting key 175E41DF from hkp server subkeys.pgp.net gpgkeys: key 175E41DF not found on keyserver gpg: no valid OpenPGP data found. gpg: Total number processed: 0 Fixed[1], seems the key has not made it to subkeys.pgp.net yet. [1] - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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
[asterisk-users] Asterisk 1.8.3.2 core dump chan_sip.c
Hello,
I'm testing with asterisk 1.8.3.2 and come across this:
Call from one extension to another with:
[macro-internal-call];ARG1=extension to call
exten = s,1,Set(TOCALL=${DB(SIP/${ARG1})})
exten = s,2,Dial(SIP/${TOCALL},60,tT)
...
As I had no entry in the asteriskdb, so the SIP uri was empty, and
asterisk core dumped with:
gdb output:
#0 0xb7c7db33 in strchr () from /lib/libc.so.6
Maybe someone can reproduce that behaviour.
yours
christian
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[asterisk-users] Updated: 10 Minutes: Asterisk PBX on Amazon EC2
Dear Asterisk Community: With more than 10,000 readers worldwide, I've refreshed my free Asterisk PBX on Amazon EC2 ebook for 2011. It has been used by Avaya, Polycom, universities, and consultants everywhere. Did I mention it's free? If you have suggestions for its improvement or things you'd like to see, please let me know! It's online here: http://ronaldlewis.com/10-minutes-asterisk-pbx-on-amazon-ec2-quickstart-guide http://www.scribd.com/doc/3905321/10-Minutes-Asterisk-PBX-on-Amazon-EC2-A-Quickstart-Guide Thanks for your support! Best, Ronald Lewis Author, 10 Minutes: Asterisk PBX on Amazon EC2 Denver, Colorado http://ronaldlewis.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dtmf_2833_1.pcap: what PCM codec? ulaw or alaw?
Hi everybody, got it from svn: dtmf_2833_1.pcap */asterisk/trunk/tests/rfc2833_dtmf_detect/configs/extensions.conf PRE-CREATION ** /asterisk/trunk/tests/rfc2833_dtmf_detect/configs/sip.conf PRE-CREATION ** /asterisk/trunk/tests/rfc2833_dtmf_detect/run-test PRE-CREATION ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/broken_dtmf.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/broken_dtmf.xml PRE-CREATION ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_1.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_1_noend.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_2.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_2_noend.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_3.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_3_noend.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_4.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_4_noend.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_5.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_6.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_7.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_8.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_9.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_pound.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_star.pcap UNKNOWN ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_baseline.xml PRE-CREATION ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_noend.xml PRE-CREATION ** /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/register.xml PRE-CREATION ** /asterisk/trunk/tests/rfc2833_dtmf_detect/test-config.yaml PRE-CREATION ** /asterisk/trunk/tests/rfc2833_dtmf_detect/test.lua PRE-CREATION * wonder what's the PCM coding format within these captures? ulaw or alaw? I'm looking for alaw captures. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Connecting Asterisk to Siemens Hipath 3750
Hi guys Thanks alot for the support. I have successfully connected the HiPath3750 to the E1 lines and everything is working fine with the appropriate dial plans. I used Josue's config and the info I got from here http://www.voip-info.org/wiki/view/Siemens+Hicom Well, not everything is working fine though.. The asterisk server seems to 'generate' the ringing tones as opposed to using the tones from the various other external numbers that I am calling. For example, if I call a phone number that is switched off, it rings for a while and then I get a service unavailable message on the IP phones. What can I do to get the normal the number you have dialed is switched off. I am in Nigeria if that information is useful in this situation. Thanks. Bobola 2011/3/16 Bobola Oke okebob...@gmail.com Hey Josue, Thanks alot. I will be expecting the configuration samples. From your response, I guess QSIG would be better for more functionality between the two PBXs then.. Yes, this is my first implementation of asterisk and the support I have had from the mailing lists (some just by searching the archives) has been nothing short of wonderful. Thanks guys. Hoping to hear from you soon. Best regards, Bobola O. Oke 2011/3/15 Josué Conti josueco...@gmail.com Hello Bobola, thanks for your response. So, I'm using Euroisdn with a Siemens HiPath 3750 and Qsig with Siemens HiPath 4000. Because we don't need to facility enable in this case (HiPath 3750) just ANI interchange between user's, ok? In another response I was send to you a configurations sample for Asterisk and Siemens may you look this? One more time, best regards and good luck in your project. If you need please contact us. Josue 2011/3/14 Bobola Oke okebob...@gmail.com Thanks guys, I got the layer1 link up. Edwin, I will make a cable from this link that you have posted and see if that also works. Presently, I just did a 'manual' connect of the ends to get the layer1 up. Josue, many thanks for your response. Searching through this list archives, I see that you must have done alot of integrating asterisk with Siemens PBX. Guys, what do you advise I use for the upper layer protocols, QSIG or EuroISDN, to connect the asterisk PBX and the Siemens PBX? What are the pros and cons of using either protocol. Working sample configuration files are highly appreciated + what the PBX guy has to configure on the Siemens side. Thanks alot. On Fri, Mar 11, 2011 at 1:17 AM, Edwin Lam edwin@officegeneral.comwrote: On 3/10/11 6:43 AM, Bobola Oke wrote: The telco has a DB9 terminated interface straight to the PBX and I cannot make sense out of the interface for the PBX. What kind of interface is this? How do I connect the RJ48 of the PRI cards to make this whole setting work. searching through this list's archive and found this: http://lists.digium.com/pipermail/asterisk-users/2006-December/174258.html -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: %2B1%20415%20439%204988 %2B1%20415%20439%204988%2B1%20415%20439%204988+1 415 439 4988 Fax: %2B1%20415%20283%203370 %2B1%20415%20283%203370%2B1%20415%20283%203370+1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Connecting Asterisk to Siemens Hipath 3750
What does your Dial command look like? If you are using the ,r option, Asterisk will generate its own ringing noise even on a dead or busy line. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bobola Oke Sent: Wednesday, March 30, 2011 11:36 AM To: Josué Conti Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Connecting Asterisk to Siemens Hipath 3750 Hi guys Thanks alot for the support. I have successfully connected the HiPath3750 to the E1 lines and everything is working fine with the appropriate dial plans. I used Josue's config and the info I got from here http://www.voip-info.org/wiki/view/Siemens+Hicom Well, not everything is working fine though.. The asterisk server seems to 'generate' the ringing tones as opposed to using the tones from the various other external numbers that I am calling. For example, if I call a phone number that is switched off, it rings for a while and then I get a service unavailable message on the IP phones. What can I do to get the normal the number you have dialed is switched off. I am in Nigeria if that information is useful in this situation. Thanks. Bobola 2011/3/16 Bobola Oke okebob...@gmail.com Hey Josue, Thanks alot. I will be expecting the configuration samples. From your response, I guess QSIG would be better for more functionality between the two PBXs then.. Yes, this is my first implementation of asterisk and the support I have had from the mailing lists (some just by searching the archives) has been nothing short of wonderful. Thanks guys. Hoping to hear from you soon. Best regards, Bobola O. Oke 2011/3/15 Josué Conti josueco...@gmail.com Hello Bobola, thanks for your response. So, I'm using Euroisdn with a Siemens HiPath 3750 and Qsig with Siemens HiPath 4000. Because we don't need to facility enable in this case (HiPath 3750) just ANI interchange between user's, ok? In another response I was send to you a configurations sample for Asterisk and Siemens may you look this? One more time, best regards and good luck in your project. If you need please contact us. Josue 2011/3/14 Bobola Oke okebob...@gmail.com Thanks guys, I got the layer1 link up. Edwin, I will make a cable from this link that you have posted and see if that also works. Presently, I just did a 'manual' connect of the ends to get the layer1 up. Josue, many thanks for your response. Searching through this list archives, I see that you must have done alot of integrating asterisk with Siemens PBX. Guys, what do you advise I use for the upper layer protocols, QSIG or EuroISDN, to connect the asterisk PBX and the Siemens PBX? What are the pros and cons of using either protocol. Working sample configuration files are highly appreciated + what the PBX guy has to configure on the Siemens side. Thanks alot. On Fri, Mar 11, 2011 at 1:17 AM, Edwin Lam edwin@officegeneral.com wrote: On 3/10/11 6:43 AM, Bobola Oke wrote: The telco has a DB9 terminated interface straight to the PBX and I cannot make sense out of the interface for the PBX. What kind of interface is this? How do I connect the RJ48 of the PRI cards to make this whole setting work. searching through this list's archive and found this: http://lists.digium.com/pipermail/asterisk-users/2006-December/174258.html -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: tel:%2B1%20415%20439%204988 tel:%2B1%20415%20439%204988 tel:%2B1%20415%20439%204988 +1 415 439 4988 tel:%2B1%20415%20439%204988 Fax: tel:%2B1%20415%20283%203370 tel:%2B1%20415%20283%203370 tel:%2B1%20415%20283%203370 +1 415 283 3370 tel:%2B1%20415%20283%203370 http://pgpkeys.mit.edu:11371/pks/lookup?op=get http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] CDR MYSQL missing field data
Warren, Thank you Eric -- Eric W. Davenport Cert-In Software Systems, Inc. P.O. Box 346 Bakersville, NC 28705 800-873-0110 ewdavenp...@certin.com www.certin.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_dahdi unknown dependency problem
So, I've compiled and installed libpri-1.4.11.5, dahdi-linux-complete-2.4.1+2.4.1 and asterisk-1.6.2.17.1, but chan_dahdi is not getting built. If I do a make menuselect in asterisk I see it listed with XXX, meaning that dependencies are not met. XXX chan_dahdi Depends on: res_smdi(M), dahdi(E), tonezone(E), pri(E), ss7(E), openr2(E) res_smdi gets built fine, dahdi is installed and working, tonezone is installed, pri is installed, ss7 is not installed, openr2 is not installed. Surely one does not need ss7 and openr2 if one has pri! So what else could be the problem? --- # ls -l /usr/lib/asterisk/modules/chan_dahdi.so ls: /usr/lib/asterisk/modules/chan_dahdi.so: No such file or directory # ls /usr/lib/asterisk/modules/res_smdi.so -l -rwxr-xr-x 1 root root 227620 Mar 30 18:35 /usr/lib/asterisk/modules/res_smdi.so # ls -l /usr/lib/libtonezone* -rwxr-xr-x 1 root root 216276 Mar 30 17:45 /usr/lib/libtonezone.a lrwxrwxrwx 1 root root 18 Mar 30 17:45 /usr/lib/libtonezone.so - libtonezone.so.2.0 lrwxrwxrwx 1 root root 18 Mar 30 17:45 /usr/lib/libtonezone.so.1 - libtonezone.so.2.0 lrwxrwxrwx 1 root root 18 Mar 30 17:45 /usr/lib/libtonezone.so.1.0 - libtonezone.so.2.0 lrwxrwxrwx 1 root root 18 Mar 30 17:45 /usr/lib/libtonezone.so.2 - libtonezone.so.2.0 -rwxr-xr-x 1 root root 214066 Mar 30 17:45 /usr/lib/libtonezone.so.2.0 # ls -l /usr/lib/libpri* -rw-r--r-- 1 root root 1224116 Mar 30 16:49 /usr/lib/libpri.a lrwxrwxrwx 1 root root 13 Mar 30 16:49 /usr/lib/libpri.so - libpri.so.1.4 -rwxr-xr-x 1 root root 790374 Mar 30 16:49 /usr/lib/libpri.so.1.4 # tail /var/log/messages Mar 30 19:24:01 stretched kernel: wct4xxp :01:02.0: RCLK source set to span 1 Mar 30 19:24:01 stretched kernel: wct4xxp :01:02.0: Recovered timing mode, RCLK set to span 1 Mar 30 19:24:01 stretched kernel: wct4xxp :01:02.0: SPAN 3: Tertiary Sync Source Mar 30 19:24:01 stretched kernel: wct4xxp :01:02.0: VPM450: Not Present Mar 30 19:24:01 stretched kernel: wct4xxp :01:02.0: TE4XXP: Span 4 configured for CCS/HDB3/CRC4 Mar 30 19:24:01 stretched kernel: wct4xxp :01:02.0: RCLK source set to span 1 Mar 30 19:24:01 stretched kernel: wct4xxp :01:02.0: Recovered timing mode, RCLK set to span 1 Mar 30 19:24:01 stretched kernel: wct4xxp :01:02.0: SPAN 4: Quaternary Sync Source Mar 30 19:24:01 stretched kernel: wct4xxp :01:02.0: VPM450: Not Present Mar 30 19:24:01 stretched dahdi: Running dahdi_cfg: succeeded Kind regards, Sebastian A -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] CDR Mysql adaptive Colum
Hello folks, i installed asterisk 1.8 from repo:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
And Looked at this article about CDR in mysl.
http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql
I installed asterisk-mysql pacakge from debian repo.
The cdr in mysql is working, but i can not get cdr adaptive colums are not,
i use this in my extension.conf
exten = s,1,set(CDR(teste)=${CHANNEL(audioreadformat)})
And is not working, i thought the only diference it i would need the colum
teste in my cdr table right ?
I tryied a lot of combinations of exten =
Does anyone have any ideia how is the right way ?
Or if i need to install anythign else to make the adaptive colums works ?
thanks!!
[]'sf.rique
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Re: [asterisk-users] asterisk and fail2ban
On Wed, 30 Mar 2011, Terry Brummell wrote: I think you will find Fail2Ban the defacto standard. I don't use fai2ban. Never have, never will because I simply don't need it. Standard iptables are good enough if you can be bothered to use them to their full abilities. No need for anything else as iptables can do connection tracking and blocking against time - just like fail2ban does. More than X connections a second/minute/hour from a given IP address? Yes, iptables can detect and block that. Works for all protocolls too - SIP, IAX, POP, SSH, etc. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 and fail2ban
could you please elaborate on how you have iptables setup to work that way? On Wed, Mar 30, 2011 at 4:11 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Wed, 30 Mar 2011, Terry Brummell wrote: I think you will find Fail2Ban the defacto standard. I don't use fai2ban. Never have, never will because I simply don't need it. Standard iptables are good enough if you can be bothered to use them to their full abilities. No need for anything else as iptables can do connection tracking and blocking against time - just like fail2ban does. More than X connections a second/minute/hour from a given IP address? Yes, iptables can detect and block that. Works for all protocolls too - SIP, IAX, POP, SSH, etc. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 and fail2ban
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, March 30, 2011 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and fail2ban could you please elaborate on how you have iptables setup to work that way? On Wed, Mar 30, 2011 at 4:11 PM, Gordon Henderson gordon+aster...@drogon.net mailto:gordon%2baster...@drogon.net wrote: On Wed, 30 Mar 2011, Terry Brummell wrote: I think you will find Fail2Ban the defacto standard. I don't use fai2ban. Never have, never will because I simply don't need it. Standard iptables are good enough if you can be bothered to use them to their full abilities. No need for anything else as iptables can do connection tracking and blocking against time - just like fail2ban does. More than X connections a second/minute/hour from a given IP address? Yes, iptables can detect and block that. Works for all protocolls too - SIP, IAX, POP, SSH, etc. Gordon -- Yah, sounds simple, how do you set it up to do this? Fail2Ban was pretty easy, if it's that easy, why was F2B even created? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 and fail2ban
I don't use F2B either, but from what I understand, it is a packaged iptables automation. If you are a unix/linux guru or have a small amount of traffic, I can see where manual iptables maintenance would be fine; F2B would be for the less-informed or more heavily attacked amongst us. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell Sent: Wednesday, March 30, 2011 3:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and fail2ban From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, March 30, 2011 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and fail2ban could you please elaborate on how you have iptables setup to work that way? On Wed, Mar 30, 2011 at 4:11 PM, Gordon Henderson gordon+aster...@drogon.net mailto:gordon%2baster...@drogon.net wrote: On Wed, 30 Mar 2011, Terry Brummell wrote: I think you will find Fail2Ban the defacto standard. I don't use fai2ban. Never have, never will because I simply don't need it. Standard iptables are good enough if you can be bothered to use them to their full abilities. No need for anything else as iptables can do connection tracking and blocking against time - just like fail2ban does. More than X connections a second/minute/hour from a given IP address? Yes, iptables can detect and block that. Works for all protocolls too - SIP, IAX, POP, SSH, etc. Gordon -- Yah, sounds simple, how do you set it up to do this? Fail2Ban was pretty easy, if it's that easy, why was F2B even created? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 and fail2ban
On Wed, Mar 30, 2011 at 03:36:10PM -0500, Danny Nicholas wrote: I don't use F2B either, but from what I understand, it is a packaged iptables automation. If you are a unix/linux guru or have a small amount of traffic, I can see where manual iptables maintenance would be fine; F2B would be for the less-informed or more heavily attacked amongst us. Fail2ban monitors log files. It looks for certain regular expressions. When those are matched frequiently enough, it runs a certain action. So in this case if it sees lines for a failed SIP registration / invite in /var/log/asterisk/messages from a certain IP address, it will add an iptables rule to block that IP address (in one specific chain). Sure, you can do that manually. Or with your own monitoring script. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtmf_2833_1.pcap: what PCM codec? ulaw or alaw?
On 03/30/2011 10:46 AM, cajsdy wrote: Hi everybody, got it from svn: dtmf_2833_1.pcap //asterisk/trunk/tests/rfc2833_dtmf_detect/configs/extensions.conf PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/configs/sip.conf PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/run-test PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/broken_dtmf.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/broken_dtmf.xml PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_1.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_1_noend.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_2.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_2_noend.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_3.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_3_noend.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_4.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_4_noend.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_5.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_6.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_7.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_8.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_9.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_pound.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_star.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_baseline.xml PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_noend.xml PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/register.xml PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/test-config.yaml PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/test.lua PRE-CREATION / wonder what's the PCM coding format within these captures? ulaw or alaw? I'm looking for alaw captures. Since they are packet captures of RTP frames, the RTP payload numbers in the frames will indicate what format they contain. Have you opened the files with Wireshark or any other tool that can interpret PCAP files? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On 03/29/2011 07:16 AM, Gilles wrote: On Mon, 28 Mar 2011 08:20:23 -0400, vip killavipki...@gmail.com wrote: Is anyone using asterisk with fail2ban? Sorry for hi-jacking the thread, but I was wondering if there were a lighter alternative that I could run on appliances? Python uses too much RAM, but I need to find a way to ban hackers from trying to connect to Asterisk from the Net. Gilles, One of our developers on the AstLinux team worked out a plugin for Arno's firewall (iptables based) which performs similar to fail2ban, but uses bash. He called it adaptive-ban. You might be able to adapt it for your use, but as it's written, it's integrated with AstLinux. http://astlinux.svn.sourceforge.net/viewvc/astlinux/branches/0.7/package/arnofw/adaptive-ban/ Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] securing sip with iptables [was: asterisk and fail2ban]
On 3/30/2011 4:25 PM, vip killa wrote: could you please elaborate on how you have iptables setup to work that way? I have my config at: http://jeremy.kister.net/code/iptables/ if you already have an iptables config and you just want to make it more secure, the magic happens in the if [ $THROTTLE ] section. if not, just: # make-non-na.pl # vi iptables ## change the MYLAN=10.0.0.0 to whatever you use ## change the RTPRANGE to whatever you have in rtp.conf # mv iptables.init /etc/init.d/iptables # /etc/init.d/iptables start -- Jeremy Kister http://jeremy.kister.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] asterisk and fail2ban
Look into the ipt_recent / xt_recent module. It's probably what he is using. On Wed, Mar 30, 2011 at 4:25 PM, vip killa vipki...@gmail.com wrote: could you please elaborate on how you have iptables setup to work that way? On Wed, Mar 30, 2011 at 4:11 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Wed, 30 Mar 2011, Terry Brummell wrote: I think you will find Fail2Ban the defacto standard. I don't use fai2ban. Never have, never will because I simply don't need it. Standard iptables are good enough if you can be bothered to use them to their full abilities. No need for anything else as iptables can do connection tracking and blocking against time - just like fail2ban does. More than X connections a second/minute/hour from a given IP address? Yes, iptables can detect and block that. Works for all protocolls too - SIP, IAX, POP, SSH, etc. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 and fail2ban
On Wed, 30 Mar 2011 16:54:51 -0500, Darrick Hartman dhart...@djhsolutions.com wrote: One of our developers on the AstLinux team worked out a plugin for Arno's firewall (iptables based) which performs similar to fail2ban, but uses bash. He called it adaptive-ban. You might be able to adapt it for your use, but as it's written, it's integrated with AstLinux. Thanks Darrick. I'll add it to the list of options to check out. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Asterisk
Kindly find below my notes preceded by ( * ). Good morning, from the last question i assume you're looking for a SIP-based configureation. On 03-30-2011 00:16, bilal ghayyad wrote: 1) How I can assign for each button an extension? you can configure them as lines (at least in my 7940). look for linex_name, linex_authname and linex_password settings in the config file. * So I will need to have a TFTP server to place the configuration file on it and to be downloaded when the Phone is booting? I can not do the configuration from the web based of the Phone? 2) How I can assign for specific button a feature to be used (like call forward or call pickup .. etc)? AFAIK you can't reprogram the softkeys. There are two buttons which you can use for programming (well, sort of). You can define the mailbox extension which can be any extension. You can write a dial plan for a specific function and then use it as voicemail. The other button is the service button which can be programmed to access any HTTP url. I'm using mine to switch my desk lamp on or off. * Can I understand that working with Asterisk does not give a chance to have IP Phones with featues assigned on the buttons, so the only way to use the features is to be by access code and can not be by button? No way to assign the access code to the button? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtmf_2833_1.pcap: what PCM codec? ulaw or alaw?
thanks. opened with wireshark and it's clearly show the format, digits. another question: each pcap has multiple frames, each frame has digit, volume 10, and duration. why it has more than one frame? would not it send multiple digit each pcap? On Wed, Mar 30, 2011 at 5:47 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 03/30/2011 10:46 AM, cajsdy wrote: Hi everybody, got it from svn: dtmf_2833_1.pcap //asterisk/trunk/tests/rfc2833_dtmf_detect/configs/extensions.conf PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/configs/sip.conf PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/run-test PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/broken_dtmf.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/broken_dtmf.xml PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_1.pcap UNKNOWN // /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_1_noend.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_2.pcap UNKNOWN // /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_2_noend.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_3.pcap UNKNOWN // /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_3_noend.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_4.pcap UNKNOWN // /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_4_noend.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_5.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_6.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_7.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_8.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_9.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_pound.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_star.pcap UNKNOWN ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_baseline.xml PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_noend.xml PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/register.xml PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/test-config.yaml PRE-CREATION ///asterisk/trunk/tests/rfc2833_dtmf_detect/test.lua PRE-CREATION / wonder what's the PCM coding format within these captures? ulaw or alaw? I'm looking for alaw captures. Since they are packet captures of RTP frames, the RTP payload numbers in the frames will indicate what format they contain. Have you opened the files with Wireshark or any other tool that can interpret PCAP files? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Asterisk
You haven't said which model Cisco phone your working with. There are several different models and they all have different configuration options. Also, asterisk in itself doesn't have anything to do with button assignments on phones. Cisco phones tend to be harder to manipulate soft-keys than say, Polycom or Aastea phones. But no matter, whichever phone you chose, you'll likely have to do any custom button assignments in the phone's config, whether that be a file or a webapp. Thanks, --Warren Selby, dCAP On Mar 30, 2011, at 6:10 PM, bilal ghayyad bilmar...@yahoo.com wrote: Kindly find below my notes preceded by ( * ). Good morning, from the last question i assume you're looking for a SIP-based configureation. On 03-30-2011 00:16, bilal ghayyad wrote: 1) How I can assign for each button an extension? you can configure them as lines (at least in my 7940). look for linex_name, linex_authname and linex_password settings in the config file. * So I will need to have a TFTP server to place the configuration file on it and to be downloaded when the Phone is booting? I can not do the configuration from the web based of the Phone? 2) How I can assign for specific button a feature to be used (like call forward or call pickup .. etc)? AFAIK you can't reprogram the softkeys. There are two buttons which you can use for programming (well, sort of). You can define the mailbox extension which can be any extension. You can write a dial plan for a specific function and then use it as voicemail. The other button is the service button which can be programmed to access any HTTP url. I'm using mine to switch my desk lamp on or off. * Can I understand that working with Asterisk does not give a chance to have IP Phones with featues assigned on the buttons, so the only way to use the features is to be by access code and can not be by button? No way to assign the access code to the button? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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
