[asterisk-users] voicemail to non-default context user does not work
Hi, I input 0203# after mailbox? voice prompt from Voicemail cmd on extensions.conf such as exten = 0021,1,Ringing exten = 0021,2,Wait(1) exten = 0021,3,Voicemail exten = 0021,4,Hangup *CLI -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/0103-09a308b0, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(SIP/0103-09a308b0, 1) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMail(SIP/0103-09a308b0, ) in new stack -- SIP/0103-09a308b0 Playing 'vm-whichbox' (language 'en') [Feb 9 17:11:54] WARNING[3574]: app_voicemail.c:2850 leave_voicemail: No entry in voicemail config file for '0203' -- Executing [EMAIL PROTECTED]:4] Hangup(SIP/0103-09a308b0, ) in new stack == Spawn extension (sip, 0021, 4) exited non-zero on 'SIP/0103-09a308b0' -- Executing [EMAIL PROTECTED]:1] Hangup(SIP/0103-09a308b0, ) in new stack == Spawn extension (sip, h, 1) exited non-zero on 'SIP/0103-09a308b0' I have an entry of 0203 at Context 03 on voicemail.conf as follows; *CLI voicemail show users ContextMbox User Zone NewMsg defaultgeneral New User 0 03 01030 03 02030 03 03030 But, I could not enter into [EMAIL PROTECTED] mailbox. Any idea? asterisk-1.4.18 -- Zen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506
Ravi, there is a easy way of creating all those commands in linux. just run the following in a shell: for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit permit udp host 192.168.5.0 eq $x any conduit permit udp host;done This will create all your PIX rules at ones. I think you could also use Cisco ACL's access-list [name] permit udp [source] [destination] range This would be in your case something like: access-list asterisk permit udp 0.0.0.0 192.168.5.0 range 1 10050 Good luck. Joris Ravichandran Rajagopal wrote: Otis, I wanted to clarify what you said and what I comprehended. the SIP protocols are disabled in fixup. Having said that I guess all I have to do is just the following. the inside IP of asterisk server is 192.168.5.0 On the cisco PIX firewall enter the following. 192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq 10002 any conduit permit udp host ... . 192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq 10050 any conduit permit udp host in the rtp.conf in /etc/asterisk change the ending port 2 (which is what it currently is) to 10050 Is there an easier way to make the entries in Cisco PIX firewall ? Thx Ravi -Original Message- From: ListAcct [mailto:[EMAIL PROTECTED] Sent: Saturday, February 09, 2008 12:18 AM To: [EMAIL PROTECTED] Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506 No problem. :-P I thought it might wise to include everything you needed just in case!! LOL! You are welcome!!! --Otis Ravichandran Rajagopal wrote: LOL I guess all I was asking for the changes to be made in the Cisco PIX 506. I think you gave me a short tutorial on VI as well. Thanks once again for this help. Let me work on these changes and test the one-way audio problem and go from there. Thx Ravi -Original Message- From: ListAcct [mailto:[EMAIL PROTECTED] Sent: Friday, February 08, 2008 11:55 PM To: [EMAIL PROTECTED] Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506 Ravi, I will explain changing the config in asterisk and the pix: Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port span to 1 to 10050 (to start, you will need to increase later as ports fill up) (use insert to make a change in a file) to save: 1. esc 2. shift + colon 3. wq (to save) If you made a mistake and do not want to save but you changed something in the file: 1. esc 2. shift + colon 3. q! (to exit) Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this case the static and conduit commands so this is a example from my setup. Theses are not usable IPs on the Internet or my IPs but just an example outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254) dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254) interface ethernet0 100full (sets the duplex and turns on interface) interface ethernet1 100full (sets the duplex and turns on interface) nameif ethernet0 outside security0 ( lower security) nameif ethernet1 dmz security50 (higher security) no fixup protocol sip 5060 no fixup protocol sip udp 5060 ! - this makes things easier so now the pix knows the IP of the asterisk box and maps the ip to the name just for configuration purposes only so if you had 20 servers or devices you wanted public access to it's just easier to remember their names versus IPs. name 192.168.254.11 dns name 192.168.254.10 asterisk ! - the static command is used as a permanent mapper from one inside, dmz, or other to the global ip vice versa. (Rule of thumb if you map using static make sure you have a conduit command) static (dmz,outside) 192.168.1.22 asterisk netmask 255.255.255.255 0 0 ! - here is where you open the ports on the global side to the asterisk box. (the conduit command allows connections from lower security interfaces to higher security interfaces) conduit permit udp host 192.168.1.22 eq 1 any conduit permit udp host 192.168.1.22 eq 10001 any conduit permit udp host 192.168.1.22 eq 10002 any conduit permit udp host 192.168.1.22 eq 10003 any conduit permit udp host 192.168.1.22 eq 10004 any conduit permit udp host 192.168.1.22 eq 10005 any Hope this helps! --Otis Ravichandran Rajagopal wrote: Otis, I am new to Cisco PIX 506 and I am learning this. If you can help me with how to do this change on Cisco PIX it would be greatly appreciated. Thx Ravi -Original Message- From: ListAcct [mailto:[EMAIL PROTECTED]
[asterisk-users] How to detect if SIP extension BUSY?
Hi, My problem is in subject. As I read in documentations and voip-info.org I can't user ChanIsAvalil because it not detects BUSY information on SIP channel. I've tried to use SIPPEER function, but it gives OK (9 ms) back on BUSY SIP channel. I use Asterisk 1.2.15, SIP extensions are Linksys PAP2. Have any other idea? -- Best regards, Csibra Gergomailto:[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] External MWI question for Asterisk
- Original Message From: Olivier [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, 9 February, 2008 6:55:15 AM Subject: Re: [asterisk-users] External MWI question for Asterisk Do you mean your script does send a NOTIFY messages to hardphones ? Then, how did you write such SIP-aware script (language, ...) ? If not, how external script and Asterisk do communicate ? Our MWI program is written in C# but to send SIP NOTIFY requests is very straight forward and could be done in anything. Our program is 250 lines of code and that includes polling the database to decide when to send the NOTIFY requests. In the NOTIFY request the body consists of: Messages-Waiting: yes|no That's all you need to turn on and off the MWI indicators on every device we've come across. There's nothing tricky to MWI at all. Regards, Greyman. Get the name you always wanted with the new y7mail email address. www.yahoo7.com.au/y7mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP user registration and Asterisk Realtime
Hi, I have installed asterisk real time and sip buddies information is being stored in DB. Now I have a question, Asterisk Realtime Server -A Third party SIP server-B Question: Is there any configuration in * RT that it can register with defined sip user on Server-B I was only able to find sip users information in DB not about user registration on other server. -ag ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor Asterisk using C
Thank you for replying. The probleam is how do I use the Asterisk_manager-API and implement them in my C code. Like how do I call a API in my C program. It will be of great help if I can have an example. By traffic I mean how much bandwidth or data transferring is taking place in a call that is network traffic. I want to monitor the asterisk server. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BLF and Asterisk 1.6.0b2
Is anyone else having trouble with Asterisk 1.6.0b2 not sending busy hints to phones? I'm not reporting this a s a bug because (although I have it working with Asterisk 1.4.17, the hardware involved is different. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco phone 79xx get database information
Hi, may be this question is a bit silly, but I couldn¹t find any document or post or anything that say that if this is possible or not. I want to show information on my phones cisco 7960/40 when a call arrive. May be a bit more than a caller ID, show more detail level, if is that possible. I already have an asterisk and the phones registered there, and I need to show on the phone display, when the call is ringing, the customer information.. Something like this: source number: xx Customer Number: x Name of Customer: x I have that information on an external database, and I know for the source number witch customer is calling. The thing is how can I show on my cisco phone the information that I have on my database when i receive a call? Do I have to configure the directory services on the phones to look for the information on the database? Can I show those lines on the phone? or I have a limit of lines or info to show? Any help would be really appreciated Thanks in advance!! cheers Javito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco phone 79xx get database information
Javier Temponi wrote: Hi, may be this question is a bit silly, but I couldn’t find any document or post or anything that say that if this is possible or not. I want to show information on my phones cisco 7960/40 when a call arrive. May be a bit more than a caller ID, show more detail level, if is that possible. I already have an asterisk and the phones registered there, and I need to show on the phone display, when the call is ringing, the customer information.. Something like this: source number: xx Customer Number: x Name of Customer: x You may want to check out this bug: http://bugs.digium.com/view.php?id=8824 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] Monitor Asterisk using C
Soumya Kat wrote: What I would like to know is how to get information such as SIP users, number of SIP connections and traffic associated with those from asterisk using a C Code. Russell Bryant There is actually no good way to do this inside of Asterisk right now. It's certainly all possible ... it's just software ... but there is no straightforward common API to accomplish what you are looking for. Well then how can I monitor asterisk using net-SNMP. There should be someway in which I can monitor asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506
I tried the following ACL command
access-list asterisk permit udp 0.0.0.0 192.168.5.0 range 1 2
and I got the following response back
[no] access-list id [line line-num] deny|permit icmp
sip smask | interface if_name | object-group
network_obj_grp_id
dip dmask | interface if_name | object-group
network_obj_grp_id
[icmp_type | object-group icmp_type_obj_grp_id]
[log [disable|default] | [level] [interval secs]]
Restricted ACLs for route-map use:
[no] access-list id deny|permit {any | prefix mask | host address}
Command failed
I don't know how to enter into the linux interface of the Cisco Pix 506
firewall
-Original Message-
From: Joris Cras [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 09, 2008 3:23 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
506
Ravi,
there is a easy way of creating all those commands in linux.
just run the following in a shell:
for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
permit udp host 192.168.5.0 eq $x any conduit permit udp host;done
This will create all your PIX rules at ones.
I think you could also use Cisco ACL's
access-list [name] permit udp [source] [destination] range
This would be in your case something like:
access-list asterisk permit udp 0.0.0.0 192.168.5.0 range 1 10050
Good luck.
Joris
Ravichandran Rajagopal wrote:
Otis,
I wanted to clarify what you said and what I comprehended.
the SIP protocols are disabled in fixup.
Having said that I guess all I have to do is just the following.
the inside IP of asterisk server is 192.168.5.0
On the cisco PIX firewall enter the following.
192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq 10001 any
conduit permit udp host
192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq 10002 any
conduit permit udp host
...
.
192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq 10050 any
conduit permit udp host
in the rtp.conf in /etc/asterisk
change the ending port 2 (which is what it currently is) to 10050
Is there an easier way to make the entries in Cisco PIX firewall ?
Thx
Ravi
-Original Message-
From: ListAcct [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 09, 2008 12:18 AM
To: [EMAIL PROTECTED]
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
506
No problem. :-P I thought it might wise to include everything you
needed just in case!! LOL! You are welcome!!!
--Otis
Ravichandran Rajagopal wrote:
LOL I guess all I was asking for the changes to be made in the Cisco PIX
506. I think you gave me a short tutorial on VI as well. Thanks once
again
for this help. Let me work on these changes and test the one-way audio
problem and go from there.
Thx
Ravi
-Original Message-
From: ListAcct [mailto:[EMAIL PROTECTED]
Sent: Friday, February 08, 2008 11:55 PM
To: [EMAIL PROTECTED]
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
506
Ravi,
I will explain changing the config in asterisk and the pix:
Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port span to
1 to 10050 (to start, you will need to increase later as ports fill
up)
(use insert to make a change in a file)
to save:
1. esc
2. shift + colon
3. wq (to save)
If you made a mistake and do not want to save but you changed something
in the file:
1. esc
2. shift + colon
3. q! (to exit)
Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this case the
static and conduit commands so this is a example from my setup.
Theses are not usable IPs on the Internet or my IPs but just an
example
outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254)
dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254)
interface ethernet0 100full (sets the duplex and turns on interface)
interface ethernet1 100full (sets the duplex and turns on interface)
nameif ethernet0 outside security0 ( lower security)
nameif ethernet1 dmz security50 (higher security)
no fixup protocol sip 5060
no fixup protocol sip udp 5060
! - this makes things easier so now the pix knows the IP of the asterisk
box and maps the ip to the name just for configuration purposes only so
if you had 20 servers or devices you wanted public access to it's just
easier to remember their names versus IPs.
name 192.168.254.11 dns
name 192.168.254.10 asterisk
! - the static command is used as a permanent mapper from one inside,
dmz, or other to the global ip vice versa. (Rule of thumb if
Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506
try:
access-list asterisk permit udp any host x.x.x.x eq 1
- Ravichandran Rajagopal [EMAIL PROTECTED] wrote:
I tried the following ACL command
access-list asterisk permit udp 0.0.0.0 192.168.5.0 range 1
2
and I got the following response back
[no] access-list id [line line-num] deny|permit icmp
sip smask | interface if_name | object-group
network_obj_grp_id
dip dmask | interface if_name | object-group
network_obj_grp_id
[icmp_type | object-group icmp_type_obj_grp_id]
[log [disable|default] | [level] [interval secs]]
Restricted ACLs for route-map use:
[no] access-list id deny|permit {any | prefix mask | host
address}
Command failed
I don't know how to enter into the linux interface of the Cisco Pix
506
firewall
-Original Message-
From: Joris Cras [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 09, 2008 3:23 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco
pix
506
Ravi,
there is a easy way of creating all those commands in linux.
just run the following in a shell:
for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
permit udp host 192.168.5.0 eq $x any conduit permit udp host;done
This will create all your PIX rules at ones.
I think you could also use Cisco ACL's
access-list [name] permit udp [source] [destination] range
This would be in your case something like:
access-list asterisk permit udp 0.0.0.0 192.168.5.0 range 1
10050
Good luck.
Joris
Ravichandran Rajagopal wrote:
Otis,
I wanted to clarify what you said and what I comprehended.
the SIP protocols are disabled in fixup.
Having said that I guess all I have to do is just the following.
the inside IP of asterisk server is 192.168.5.0
On the cisco PIX firewall enter the following.
192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq
10001 any
conduit permit udp host
192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq
10002 any
conduit permit udp host
...
.
192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq
10050 any
conduit permit udp host
in the rtp.conf in /etc/asterisk
change the ending port 2 (which is what it currently is) to
10050
Is there an easier way to make the entries in Cisco PIX firewall ?
Thx
Ravi
-Original Message-
From: ListAcct [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 09, 2008 12:18 AM
To: [EMAIL PROTECTED]
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] oneway audio with asterisk behind
cisco pix
506
No problem. :-P I thought it might wise to include everything you
needed just in case!! LOL! You are welcome!!!
--Otis
Ravichandran Rajagopal wrote:
LOL I guess all I was asking for the changes to be made in the
Cisco PIX
506. I think you gave me a short tutorial on VI as well. Thanks
once
again
for this help. Let me work on these changes and test the one-way
audio
problem and go from there.
Thx
Ravi
-Original Message-
From: ListAcct [mailto:[EMAIL PROTECTED]
Sent: Friday, February 08, 2008 11:55 PM
To: [EMAIL PROTECTED]
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] oneway audio with asterisk behind
cisco pix
506
Ravi,
I will explain changing the config in asterisk and the pix:
Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port
span to
1 to 10050 (to start, you will need to increase later as ports
fill
up)
(use insert to make a change in a file)
to save:
1. esc
2. shift + colon
3. wq (to save)
If you made a mistake and do not want to save but you changed
something
in the file:
1. esc
2. shift + colon
3. q! (to exit)
Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this
case the
static and conduit commands so this is a example from my setup.
Theses are not usable IPs on the Internet or my IPs but just an
example
outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254)
dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254)
interface ethernet0 100full (sets the duplex and turns on
interface)
interface ethernet1 100full (sets the duplex and turns on
interface)
nameif ethernet0 outside security0 ( lower security)
nameif ethernet1 dmz security50 (higher security)
no fixup protocol sip 5060
no fixup protocol sip udp 5060
! - this makes things easier so now the pix knows the IP of the
asterisk
box and maps the ip to the name just for configuration purposes
only so
if you had 20
Re: [asterisk-users] Upgrade 1.2 - 1.4 voice files
Adrian Marsh wrote: In the Make menuselect, I noticed theres no .SLN voicefile selection for the basic audiofiles - has SLN been depreciated? No, the sln format is still supported. We have just never distributed any files in that raw format. Previously, we only had gsm recordings. For Asterisk 1.4, we got all of the prompts re-recorded so that we could distribute them in a number of higher-quality codecs, as well as in 3 languages. The actually scripts of the files has not changed much, as far as I remember. The sounds.txt file in 1.2, and the 1.4 sounds packages should say exactly what they are. You can always compare them with diff. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF and Asterisk 1.6.0b2
Thomas Kenyon wrote: Is anyone else having trouble with Asterisk 1.6.0b2 not sending busy hints to phones? I'm not reporting this a s a bug because (although I have it working with Asterisk 1.4.17, the hardware involved is different. What type of device are you subscribing to, is it another SIP phone? If so, what is the associated configuration in sip.conf? Do you have call-limit set to some value, or the combination of callcounter and busylevel? If so, what are they set to? (You must have these options set for it to work) -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HP proliant and hpasm
Is anyone successfully running asterisk on an HP proliant while using their management software, hpasm? I have two DL360's and two TE220B's. The cards have their own IRQ's. No matter what combination of settings I use, the cards fail the patlooptest if hpasm (ver 7.9.1) is running. If I stop it the cards pass the test. I really want to run the management software, so I'd like to know if anyone has it working. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF and Asterisk 1.6.0b2
Russell Bryant wrote: Thomas Kenyon wrote: Is anyone else having trouble with Asterisk 1.6.0b2 not sending busy hints to phones? I'm not reporting this a s a bug because (although I have it working with Asterisk 1.4.17, the hardware involved is different. What type of device are you subscribing to, is it another SIP phone? If so, what is the associated configuration in sip.conf? Do you have call-limit set to some value, or the combination of callcounter and busylevel? If so, what are they set to? (You must have these options set for it to work) I have enough kit around to set the machine I'm testing 1.6.0b2 to use the same configuration as the working machines. I have got call-limits set, but it did occur to me that there's no reason asterisk would know that there is only one extension on SIP/peername. The stranger thing is, on the machine that it's all working on, there is a call-limit=4 set on every extension (from what I remember it prevented a bug that got fixed ages ago and I didn't get round to lowering it again). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506
I made the following changes and I am still facing one way audio with my call
flow.
-Original Message-
From: Wendell Hamilton [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 09, 2008 1:58 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Joris Cras; [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506
try:
access-list asterisk permit udp any host x.x.x.x eq 1
- Ravichandran Rajagopal [EMAIL PROTECTED] wrote:
I tried the following ACL command
access-list asterisk permit udp 0.0.0.0 192.168.5.0 range 1
2
and I got the following response back
[no] access-list id [line line-num] deny|permit icmp
sip smask | interface if_name | object-group
network_obj_grp_id
dip dmask | interface if_name | object-group
network_obj_grp_id
[icmp_type | object-group icmp_type_obj_grp_id]
[log [disable|default] | [level] [interval secs]]
Restricted ACLs for route-map use:
[no] access-list id deny|permit {any | prefix mask | host
address}
Command failed
I don't know how to enter into the linux interface of the Cisco Pix
506
firewall
-Original Message-
From: Joris Cras [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 09, 2008 3:23 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco
pix
506
Ravi,
there is a easy way of creating all those commands in linux.
just run the following in a shell:
for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
permit udp host 192.168.5.0 eq $x any conduit permit udp host;done
This will create all your PIX rules at ones.
I think you could also use Cisco ACL's
access-list [name] permit udp [source] [destination] range
This would be in your case something like:
access-list asterisk permit udp 0.0.0.0 192.168.5.0 range 1
10050
Good luck.
Joris
Ravichandran Rajagopal wrote:
Otis,
I wanted to clarify what you said and what I comprehended.
the SIP protocols are disabled in fixup.
Having said that I guess all I have to do is just the following.
the inside IP of asterisk server is 192.168.5.0
On the cisco PIX firewall enter the following.
192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq
10001 any
conduit permit udp host
192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq
10002 any
conduit permit udp host
...
.
192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq
10050 any
conduit permit udp host
in the rtp.conf in /etc/asterisk
change the ending port 2 (which is what it currently is) to
10050
Is there an easier way to make the entries in Cisco PIX firewall ?
Thx
Ravi
-Original Message-
From: ListAcct [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 09, 2008 12:18 AM
To: [EMAIL PROTECTED]
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] oneway audio with asterisk behind
cisco pix
506
No problem. :-P I thought it might wise to include everything you
needed just in case!! LOL! You are welcome!!!
--Otis
Ravichandran Rajagopal wrote:
LOL I guess all I was asking for the changes to be made in the
Cisco PIX
506. I think you gave me a short tutorial on VI as well. Thanks
once
again
for this help. Let me work on these changes and test the one-way
audio
problem and go from there.
Thx
Ravi
-Original Message-
From: ListAcct [mailto:[EMAIL PROTECTED]
Sent: Friday, February 08, 2008 11:55 PM
To: [EMAIL PROTECTED]
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] oneway audio with asterisk behind
cisco pix
506
Ravi,
I will explain changing the config in asterisk and the pix:
Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port
span to
1 to 10050 (to start, you will need to increase later as ports
fill
up)
(use insert to make a change in a file)
to save:
1. esc
2. shift + colon
3. wq (to save)
If you made a mistake and do not want to save but you changed
something
in the file:
1. esc
2. shift + colon
3. q! (to exit)
Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this
case the
static and conduit commands so this is a example from my setup.
Theses are not usable IPs on the Internet or my IPs but just an
example
outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254)
dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254)
interface ethernet0 100full (sets
[asterisk-users] Carrier SIP resource?
Dear List: Can anyone refer me to a resource to better understand how the SIP protocol is used by carriers to provide voice circuits between * and the PSTN? Thanks a bunch! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506
Did you only open up the one port (1)? You need to open up a range, if
you're doing it this way, like 1-10020 and then set your rtp ports in
asterisk to the same range.
- Ravichandran Rajagopal [EMAIL PROTECTED] wrote:
I made the following changes and I am still facing one way audio with
my call flow.
-Original Message-
From: Wendell Hamilton [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 09, 2008 1:58 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Cc: Joris Cras; [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco
pix 506
try:
access-list asterisk permit udp any host x.x.x.x eq 1
- Ravichandran Rajagopal [EMAIL PROTECTED]
wrote:
I tried the following ACL command
access-list asterisk permit udp 0.0.0.0 192.168.5.0 range 1
2
and I got the following response back
[no] access-list id [line line-num] deny|permit icmp
sip smask | interface if_name | object-group
network_obj_grp_id
dip dmask | interface if_name | object-group
network_obj_grp_id
[icmp_type | object-group icmp_type_obj_grp_id]
[log [disable|default] | [level] [interval secs]]
Restricted ACLs for route-map use:
[no] access-list id deny|permit {any | prefix mask | host
address}
Command failed
I don't know how to enter into the linux interface of the Cisco Pix
506
firewall
-Original Message-
From: Joris Cras [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 09, 2008 3:23 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
Discussion
Subject: Re: [asterisk-users] oneway audio with asterisk behind
cisco
pix
506
Ravi,
there is a easy way of creating all those commands in linux.
just run the following in a shell:
for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
permit udp host 192.168.5.0 eq $x any conduit permit udp host;done
This will create all your PIX rules at ones.
I think you could also use Cisco ACL's
access-list [name] permit udp [source] [destination] range
This would be in your case something like:
access-list asterisk permit udp 0.0.0.0 192.168.5.0 range 1
10050
Good luck.
Joris
Ravichandran Rajagopal wrote:
Otis,
I wanted to clarify what you said and what I comprehended.
the SIP protocols are disabled in fixup.
Having said that I guess all I have to do is just the following.
the inside IP of asterisk server is 192.168.5.0
On the cisco PIX firewall enter the following.
192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq
10001 any
conduit permit udp host
192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq
10002 any
conduit permit udp host
...
.
192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq
10050 any
conduit permit udp host
in the rtp.conf in /etc/asterisk
change the ending port 2 (which is what it currently is) to
10050
Is there an easier way to make the entries in Cisco PIX firewall
?
Thx
Ravi
-Original Message-
From: ListAcct [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 09, 2008 12:18 AM
To: [EMAIL PROTECTED]
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] oneway audio with asterisk behind
cisco pix
506
No problem. :-P I thought it might wise to include everything
you
needed just in case!! LOL! You are welcome!!!
--Otis
Ravichandran Rajagopal wrote:
LOL I guess all I was asking for the changes to be made in the
Cisco PIX
506. I think you gave me a short tutorial on VI as well. Thanks
once
again
for this help. Let me work on these changes and test the one-way
audio
problem and go from there.
Thx
Ravi
-Original Message-
From: ListAcct [mailto:[EMAIL PROTECTED]
Sent: Friday, February 08, 2008 11:55 PM
To: [EMAIL PROTECTED]
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] oneway audio with asterisk behind
cisco pix
506
Ravi,
I will explain changing the config in asterisk and the pix:
Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port
span to
1 to 10050 (to start, you will need to increase later as
ports
fill
up)
(use insert to make a change in a file)
to save:
1. esc
2. shift + colon
3. wq (to save)
If you made a mistake and do not want to save but you changed
something
in the file:
1. esc
2. shift + colon
3. q! (to exit)
Cisco Pix -
Re: [asterisk-users] voicemail to non-default context user does not work
According to voip-info, the syntax for the VoiceMail command is as follows... VoiceMail([/flags/]/[EMAIL PROTECTED][EMAIL PROTECTED]boxnumber3]/) If you check the syntax for the VoiceMail command, it indicates that the mailbox parameter is /not/ optional, so I'm surprised this works at all. Asterisk will default to the @default context if the context isn't specified, so you /might/ try Voicemail(@03) otherwise I suspect you're going to need an IVR to achieve what you want. Zen Kato wrote: Hi, I input 0203# after mailbox? voice prompt from Voicemail cmd on extensions.conf such as exten = 0021,1,Ringing exten = 0021,2,Wait(1) exten = 0021,3,Voicemail exten = 0021,4,Hangup *CLI -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/0103-09a308b0, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(SIP/0103-09a308b0, 1) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMail(SIP/0103-09a308b0, ) in new stack -- SIP/0103-09a308b0 Playing 'vm-whichbox' (language 'en') [Feb 9 17:11:54] WARNING[3574]: app_voicemail.c:2850 leave_voicemail: No entry in voicemail config file for '0203' -- Executing [EMAIL PROTECTED]:4] Hangup(SIP/0103-09a308b0, ) in new stack == Spawn extension (sip, 0021, 4) exited non-zero on 'SIP/0103-09a308b0' -- Executing [EMAIL PROTECTED]:1] Hangup(SIP/0103-09a308b0, ) in new stack == Spawn extension (sip, h, 1) exited non-zero on 'SIP/0103-09a308b0' I have an entry of 0203 at Context 03 on voicemail.conf as follows; *CLI voicemail show users ContextMbox User Zone NewMsg defaultgeneral New User 0 03 01030 03 02030 03 03030 But, I could not enter into [EMAIL PROTECTED] mailbox. Any idea? asterisk-1.4.18 -- Zen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialing SIP server user extension... Dial string issue...
Why are you specifying the password and server IP in the dial string when it's included in sip.conf? It's unnecessary. I believe that Dial(SIP/gs102/1234) will achieve what you want. ast guy wrote: Hi, I'm trying to call a SIP server while providing the SIP server username/password in dial string but it's not working ... Dial(SIP/gs102:[EMAIL PROTECTED] mailto:SIP/gs102:[EMAIL PROTECTED]); User on sip server (192.168.2.81 http://192.168.2.81): [gs102] disallow=all allow=ulaw allow=alaw type=friend username=gs102 secret=test host=dynamic dtmfmode=inband defaultip=192.168.2.1 http://192.168.2.1 qualify=1000 mailbox=102 context=context-gs102 Extensions.conf entry [context-gs102] exten = s,1, Answer(); exten = s,n, Playback(demo-congrats); exten = s,n, Meetme(8600051); exten = 1234,1, Answer(); exten = 1234,n, Playback(demo-congrats); exten = 1234,n, Meetme(8600051); When I dial I get following error on console -- Executing Dial(SIP/331-6263, SIP/gs102:[EMAIL PROTECTED] mailto:SIP/gs102:[EMAIL PROTECTED]) in new stack -- Called gs102:[EMAIL PROTECTED] mailto:gs102:[EMAIL PROTECTED] -- SIP/192.168.2.81-0343 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/331-6263, ) in new stack == Spawn extension (default, 1234, 2) exited non-zero on 'SIP/331-6263' I want to call extension 1234 defined under gs102 defined context-gs102 context... what should be the exact Dialed SIP URL ? -ag ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disappearing B-Channels
In my efforts to solve a mystery of asterisk slowly loosing it's ability to take incoming and outgoing calls I set asterisk to restart b-channels every 60 seconds hoping I would find something odd after some time. So now I am looking at the CLI a few hours later and look what happens when asterisk restarts the 23 b-channels I have. pbx1*CLI -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 == Primary D-Channel on span 1 down [Feb 10 01:41:23] WARNING[4102]: chan_zap.c:2401 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! [Feb 10 01:41:24] ERROR[4102]: chan_zap.c:8200 zt_pri_error: !! Got S-frame while link down == Primary D-Channel on span 1 up -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 -- B-channel 0/23 successfully restarted on span 1 pbx1*CLI That's the output while I've been writing this email. Those are TWO restarts of the b-channels. Notice I am missing a seizable amount of my 23 b-channels. Where are they going?! How do I find out? I've recompiled my asterisk, zaptel, and libpri to the most recent versions but that's made no difference. - Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
