[asterisk-users] Call status register
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! Some time ago I'm using Asterisk (currently 1.8.10.0) at home to manage the calls. Nothing yet very complex, just something compiled by me using the source code from the official site of the project and configuring the files manually to both Asterisk and DAHDI. For now I'm not using any GUI, but when I have more time, I plan to try something in the future, for example, to make a statistic of the calls. But, thinking about the statistics of the calls, in the last days I was taking a look at the /var/log/asterisk/cdr-csv/Master.csv file, which I understand is where the calls are registered. But all seem to have a ANSWERED state, even those receiving a busy tone. This happens with both internal calls between SIP extension and from SIP to PSTN. A test I did is putting a Grandstream BT200 on DND mode (Do Not Disturb) and call it from a softphone. While the softphone receives the message that the extension is busy, the CDR registered the call as ANSWERED. Not sure if it's something usually due to the way it is configured the dialplan or any other configuration issue. Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAk+LIeAACgkQZpa/GxTmHTcvQwCdEqsEI3Y9ka5Z41CTXlzerPbD qQIAnAwEpac8dcLh5t84XLuDryqFuD40 =R07r -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Extensions and loss of Internet connection
Hi, Phil. A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. I'm using Asterisk 1.4.24.1. I was researching on the Internet and I found that it can be related to a bug of chan_sip, can it be? In this case, is there a possible workaround? Does you Asterisk server point to an internal DNS or to your router ? The /etc/resolv.conf of the host on which I installed Asterisk points to an internal DNS. Is there a parameter in the Asterisk configuration where also I have to force the use of an internal DNS server? Thanks for your reply. Regards, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 Extensions and loss of Internet connection
Does you Asterisk server point to an internal DNS or to your router ? The /etc/resolv.conf of the host on which I installed Asterisk points to an internal DNS. Is there a parameter in the Asterisk configuration where also I have to force the use of an internal DNS server? Do your SIP extensions use your internal DNS server ? are they able to resolve the IP of your Asterisk server ? If you enable SIP debugging do you see them even try and connect ? The extensions have configured the Asterisk server by its IP, so I do not think there is a problem on that side. To enable debug I should use 'sip set debug'? from the Asterisk CLI? I do not see any record in the CLI after running this command. However, from Twinkle, for example, I see the following: - lun 10:49:59 Daniel, registration failed: 503 Service Unavailable - Thanks for your reply. Regards, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 Extensions and loss of Internet connection
Hi, Alejandro. A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. You have to be a bit more specific. For example is your Asterisk box behind a router/nat? Or does your asterisk box have two NICs one for the public and/or natted IP and one for the LAN? You need to specify your exact setup. Asterisk is not behind the router. The problem I'm having is in the LAN. As I told Phil, I am experiencing the same problem both from a softphone on a workstation with fixed IP as a Grandstream phone (which gets network configuration via DHCP). In both extensions, the Asterisk server is configured with IP, so in that sense, I don't think the server is inaccessible to customers. On the other hand, I made sure to have commented in the sip.conf file any reference to providers such as Ekiga or iptel, so the server should not be trying to get to the Internet. It would appear that the server for some reason was 'locked'. For example, when I try to register from Twinkle softphone, I get the following: - lun 13:41:56 Daniel, registration failed: 503 Service Unavailable - Thanks for your reply. Regards, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] SIP Extensions and loss of Internet connection
Hi all! A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. I'm using Asterisk 1.4.24.1. I was researching on the Internet and I found that it can be related to a bug of chan_sip, can it be? In this case, is there a possible workaround? Thanks in advance for your reply. Regards, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] About Sangoma cards and Asterisk integration with other PBX
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wed, Jun 02, 2010 at 21:50:43 -0300, Daniel Bareiro wrote: Another thing I want to try is to connect Asterisk with Siemens PBX so that the extensions on Asterisk can communicate with the extensions on the Siemens PBX and vice versa. For this should I connect a FXO channel on Asterisk with a FXS channel of Siemens PBX? That might be one way, though I would think, depending on the Siemens hardware, a T1 connection might be more flexible and provide better integration. Lamentably, for the present, I do not believe that we buy a T1 card for Asterisk. As I said in another message of this thread, when trying to communicate with an extension of the Siemens PBX, I obtain busy/congested. When searching on the Internet if Asterisk requires some special configuration to interact with this type of PBX, I found that some Siemens models use proprietary protocols [1], although I'm not sure if the problem I'm having is because of it. Our PBX has two parts. I have understood that the smallest box (than it is on the other) is the DISA. If it serves as something, in the later part it has model 7655. An additional information that I got is that Siemens PBX is Hicom 150. Have you had (or someone on the list) the opportunity to integrate this type of PBX with Asterisk through a analog card? Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkwH5ywACgkQZpa/GxTmHTer4QCeNdXMum9GU+mOAGCcFkFw5WaL WDMAn1UXK4AKyJKxzx5y4b/Em8tQLiZj =GDSS -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About Sangoma cards and Asterisk integration with other PBX
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, John. On Fri, May 21, 2010 at 23:35:41 -0300, John Novack wrote: Another thing I want to try is to connect Asterisk with Siemens PBX so that the extensions on Asterisk can communicate with the extensions on the Siemens PBX and vice versa. For this should I connect a FXO channel on Asterisk with a FXS channel of Siemens PBX? That might be one way, though I would think, depending on the Siemens hardware, a T1 connection might be more flexible and provide better integration. Lamentably, for the present, I do not believe that we buy a T1 card for Asterisk. As I said in another message of this thread, when trying to communicate with an extension of the Siemens PBX, I obtain busy/congested. When searching on the Internet if Asterisk requires some special configuration to interact with this type of PBX, I found that some Siemens models use proprietary protocols [1], although I'm not sure if the problem I'm having is because of it. Our PBX has two parts. I have understood that the smallest box (than it is on the other) is the DISA. If it serves as something, in the later part it has model 7655. I noticed that, unlike OpenVox A400P card, RJ connectors on the Sangoma A200 card are smaller. Correct. They are NOT RJ connectors, but 4 position 4 pin modular sockets, as used on US handsets. A better choice, IMO, as the 6 position 4 pin modular sockets can have the release tangs easily caught in the slot. A200 cards are provided when new, with adapter cords that have 4 position sockets on one end and 6 position on the other. Apparently, the OpenVox use standard telephone connectors. As do the Digium cards. NOTE: Using the RJ designation is not correct, though it is widely misused. RJ is an FCC designation for Registered Jack, and refers to the wiring scheme for various interconnections to the public switched network. there are 4 position, 6 position 8 position, and seldom seen 10 position modular plugs and sockets. The 4 position was only used, other than the Sangoma A200, for handsets on modular telephones, and never for PSTN connection, and never had an RJ designation. Misinformation available on the Internet shows various designations. Thanks for the explanation and clarification of nomenclature. And in what cases it would be correct to use the RJ designation? Thanks for your reply. Regards, Daniel [1] http://www.voip-info.org/wiki/view/Siemens+Hicom -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkwG/KgACgkQZpa/GxTmHTdPzgCfc7FtZPSd34tpOC9YNp64ITgw M6wAnRoE2i16KNtN0JUGyizW5eIuam4O =G2xi -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About Sangoma cards and Asterisk integration with other PBX
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Gopalakrishnan. On Fri, May 28, 2010 at 01:44:41 -0300, Gopalakrishnan A.N wrote: I suspect the channel is not ceased correctly in Siemens PBX, since you get dial tone from Siemens PBX the channel from Asterisk is rejected in your Siemens PBX. H... but this is something that should be reviewed on the side of Siemens PBX? Because I had thought it might be due to a configuration issue in Asterisk FXO channel. The strange thing is that when I connect a phone to that extension of the Siemens PBX, I get dial tone and I can even call to another extension of the Siemens PBX. In fact, callerid in the destination extension indicates that the call comes from the extension 568. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkv/hT4ACgkQZpa/GxTmHTdAJQCgjz4urW3MW5Hcpcu6c0PGaLV0 DhkAn0lpaeYjym8mMrVw65g62EJ1O6O2 =i3xV -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About Sangoma cards and Asterisk integration with other PBX
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Fri, May 21, 2010 at 23:35:14 -0300, Tim Nelson wrote:
Greetings!
Hi, Tim!
I had the opportunity to test a Sangoma A200 card and I have some
doubts that I would like to consult:
I tried to detect the card and I had no success using the wctdm
module with DAHDI. I guess this is because electronics is different
because the TDM400P and OpenVox A400P cards have separate modules for
each channel, while the Sangoma A200 each module operates two
channels. I had to compile Wanpipe for the card was detected. Is it
the only way?
Wanpipe is what interfaces the hardware with Dahdi/Zaptel. Then,
Dahdi/Zaptel interfaces with Asterisk. This is normal.
Well, then wanpipe is necessary.
Another thing I want to try is to connect Asterisk with Siemens PBX
so that the extensions on Asterisk can communicate with the
extensions on the Siemens PBX and vice versa. For this should I
connect a FXO channel on Asterisk with a FXS channel of Siemens PBX?
Personally, if possible, I'd connect one of each(FXO/FXS) on Asterisk
to one of each(FXO/FXS) on the Siemens. This allows for proper dialing
between systems and passing your ${EXTEN} as expected.
I'm not sure if I understood well. Must I use two FXO/FXS connections? A
FXO (Asterisk) / FXS (Siemens) connection and another FXO (Siemens) /
FXS (Asterisk) connection? does not serve a single connection for
incoming and outgoing calls like when we connect Asterisk to the PSTN?
I noticed that, unlike OpenVox A400P card, RJ connectors on the
Sangoma A200 card are smaller. Apparently, the OpenVox use standard
telephone connectors.
Sangoma's cards come with a half-height PCI bracket for smaller
systems. To ensure the card stays small, they use smaller jacks, RJ14
or 'handset' jacks IIRC. Again, this is something specific to Sangoma
and normal.
Today I was doing tests connecting FXO channel on Sangoma card to a
extension of Siemens PBX. Previously, connecting a phone, I made sure in
that socket I had a dial tone.
I tried calling the extension 509 on Siemens PBX, but I get a busy tone
with the following message in the CLI:
- -
dynatac*CLI
-- Executing [9...@from-internal:1] Dial(SIP/200-0004,
DAHDI/3/509) in new stack
[May 26 14:47:59] WARNING[3031]: app_dial.c:1298 dial_exec_full: Unable
to create channel of type 'DAHDI' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [9...@from-internal:2] Hangup(SIP/200-0004, )
in new stack
== Spawn extension (from-internal, 9509, 2) exited non-zero on
'SIP/200-0004'
-- Executing [9...@from-internal:1] Dial(SIP/200-0005,
DAHDI/3/509) in new stack
[May 26 14:48:32] WARNING[3032]: app_dial.c:1298 dial_exec_full: Unable
to create channel of type 'DAHDI' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [9...@from-internal:2] Hangup(SIP/200-0005, )
in new stack
== Spawn extension (from-internal, 9509, 2) exited non-zero on
'SIP/200-0005'
- -
This is the configuration I'm using in chan_dahdi.conf:
- -
[trunkgroups]
[channels]
language=es
defaultzone=es
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
inmediate=no
; DGB - 20100322
busydetect=yes
busycount=3
;Sangoma AFT-A200 [slot:8 bus:1 span:1] wanpipe1
context=from-internal
mailbox=...@voicemail
callerid=Jane Doe 300
group=1
echocancel=yes
signalling = fxo_ls
channel = 1
context=from-internal
group=2
echocancel=yes
signalling = fxo_ks
channel = 2
context=from-zaptel
group=3
echocancel=yes
signalling = fxs_ks
channel = 3
context=from-zaptel
group=4
echocancel=yes
signalling = fxs_ks
channel = 4
-
[asterisk-users] About Sangoma cards and Asterisk integration with other PBX
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I had the opportunity to test a Sangoma A200 card and I have some doubts that I would like to consult: I tried to detect the card and I had no success using the wctdm module with DAHDI. I guess this is because electronics is different because the TDM400P and OpenVox A400P cards have separate modules for each channel, while the Sangoma A200 each module operates two channels. I had to compile Wanpipe for the card was detected. Is it the only way? Another thing I want to try is to connect Asterisk with Siemens PBX so that the extensions on Asterisk can communicate with the extensions on the Siemens PBX and vice versa. For this should I connect a FXO channel on Asterisk with a FXS channel of Siemens PBX? I noticed that, unlike OpenVox A400P card, RJ connectors on the Sangoma A200 card are smaller. Apparently, the OpenVox use standard telephone connectors. Thanks in advance for your replies. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkv3NNIACgkQZpa/GxTmHTdwTQCfaVv5FZc3T33++JaiVAkgnITs vzYAnicGq+ItJH1tLYf0xMuX/peJjQxe =WVug -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callerid with DAHDI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Tzafrir. On Thu, May 20, 2010 at 09:58:26 -0300, Tzafrir Cohen wrote: I'm testing a telephone connected to FXS port of a Sangoma A200 card. But I'm observing that callerid is not working. The configuration that I'm using in chan_dahdi.conf is the following one: - - ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2010-05-11 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] language=es defaultzone=es usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes inmediate=no ; DGB - 20100322 busydetect=yes busycount=3 ;Sangoma AFT-A200 [slot:8 bus:1 span:1] wanpipe1 context=from-internal group=1 echocancel=yes signalling = fxo_ks channel = 1 mailbox=...@voicemail callerid=Jane Doe 300 The 'mailbox' and 'callerid' settings only affect channels 2-4, and not channel 1. Is that intentoinal? Hmmm ... then I just found out mailbox and callerid apply from where I put it and that its location is not trivial. Should these be over channel? Putting both under context I see that it works. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkv1ZSYACgkQZpa/GxTmHTfHwACeKO9EHjtIoe+A5/UP+/KntPwg thoAn1EQMfSksCMgqSUUh63SNYlIjanX =wFrb -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Callerid with DAHDI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I'm testing a telephone connected to FXS port of a Sangoma A200 card. But I'm observing that callerid is not working. The configuration that I'm using in chan_dahdi.conf is the following one: - - ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2010-05-11 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] language=es defaultzone=es usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes inmediate=no ; DGB - 20100322 busydetect=yes busycount=3 ;Sangoma AFT-A200 [slot:8 bus:1 span:1] wanpipe1 context=from-internal group=1 echocancel=yes signalling = fxo_ks channel = 1 mailbox=...@voicemail callerid=Jane Doe 300 context=from-internal group=1 echocancel=yes signalling = fxo_ks channel = 2 context=from-zaptel group=0 echocancel=yes signalling = fxs_ks channel = 3 context=from-zaptel group=0 echocancel=yes signalling = fxs_ks channel = 4 - - I was comparing this configuration with which I have in my house with a OpenVox card, where callerid works, and the unique difference that I found is that I'm using fxo_ls. Can be it the cause of the problem? Thanks in advance for your replies. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkvx7OkACgkQZpa/GxTmHTcWsACglvrpdwGsyzc1ZNOIJplgctsh hGEAnjEJPReISMuPUm96gX2Yqcg9WoqA =AeXz -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with Music on hold
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! During tests with a Grandstream GXP280 phone, I found that pressing 'hold' button, the other extension (Qutecom softphone) is put on hold but without music. Then, when resuming the conversation, I listen the other user again but he/her no longer listen to me. When from softphone the same test is realised, it does not happen this problem. Can it be due to a configuration problem of the Grandstream phone? Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkvuBdQACgkQZpa/GxTmHTdC8QCgilVwAkPv7VIS6AOA7pzGKmgQ EYcAnRxcvcFHAAwsVZ+Vg2ukWoPSWzmL =xdok -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security tests
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Steve. On Fri, Apr 23, 2010 at 22:38:49 -0300, Steve Totaro wrote: Perhaps it was not very clear, but yes, I was talking about this. I believe that I found the cause of the problem. The cause by which I was not seeing VoIP traffic between 10.1.0.38 and 10.1.0.65 is because there is no direct traffic among them but that is between each party and the Asterisk server :-) So using ettercap with de IP of Asterisk server and 10.1.0.65 I can now capture and play calls from this IP to 10.1.0.38 or vice versa. But I'm noticing that playing from Wireshark it can be heard delayed. Is it normal to happen? On the other hand, I had to change the order of preference of the codecs in the sip.conf so that G711 is preferred over GSM, because it was configured in a reverse order of preference and I see that the RTP player of Wireshark does not support GSM. Do you know any way to play GSM directly from the captured packets? How did you place your virtual listening machine into the network, is it connected to an old hub, or a switch, or the mirroring port of a switch, or does it use the same NIC (and computer) as the softphone? You will first need to get in between the two endpoints in order to be able to capture that point-to-point RTP traffic - there are normal and malicious ways to achieve that. I have a switch that connects to the phone (10.1.0.38), PC with softphone (10.1.0.65), the Asterisk server and a VMHost that has the virtual machine where I use ettercap and tcpdump. Check out *Cain* *Abel* http://www.oxid.it/ and OrecX http://www.orecx.com/web/products-orekagpl.php. Oreca will run just fine on your Asterisk box. I had read something about Cain Abel. I will try reproducing the capture in an equipment with Windows using Cain Abel because here, in my house, I only have GNU/Linux and OpenBSD. About the delayed reproduction on Wireshark, is it something that also you have experimented? I am not sure what kind of security audit you are trying to do. What you propose is simple and simply the way things work, it is not security. This is initially for an presentation about security in the course of Distributed Systems. My idea was to speak on attacks and countermeasures in VoIP. On the other hand, they are asking to me to make a practical demonstration of the countermeasures. Although a direct form to avoid this is using VLANs, it seems that the idea is to demonstrate the countermeasures with some software. Then I was thinking about trying, for example, SRTP or SIP over TCP/TLS. Do you have implemented it on Asterisk 1.4? In such case, could you recommend some good document on this matter? I'm using at the moment Asterisk 1.4.24.1. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkvdgVkACgkQZpa/GxTmHTfukwCgg3hf2mBvZHXqiEjk2JAvI1dW +6sAoI/bDWWfEeWvY9InSO1Pi0381uNu =hHoH -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security tests
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 El jueves 22 de abril del 2010 a las 14:33:01 -0300, Philipp von Klitzing escribió: Hi! Hi, Philipp. But it draws attention to me between the PC with softphone and the telephone I see traffic ARP or ICMP that could make to try between the equipment but does not see RTP. Is there some special consideration that it must to observe? Your English is seriously twisted, making your question impossible to understand. My feeling is that you have used a machine translation service. Your question is probably: I can see ARP and ICMP, but not RTP, what am I missing? Perhaps it was not very clear, but yes, I was talking about this. I believe that I found the cause of the problem. The cause by which I was not seeing VoIP traffic between 10.1.0.38 and 10.1.0.65 is because there is no direct traffic among them but that is between each party and the Asterisk server :-) So using ettercap with de IP of Asterisk server and 10.1.0.65 I can now capture and play calls from this IP to 10.1.0.38 or vice versa. But I'm noticing that playing from Wireshark it can be heard delayed. Is it normal to happen? On the other hand, I had to change the order of preference of the codecs in the sip.conf so that G711 is preferred over GSM, because it was configured in a reverse order of preference and I see that the RTP player of Wireshark does not support GSM. Do you know any way to play GSM directly from the captured packets? How did you place your virtual listening machine into the network, is it connected to an old hub, or a switch, or the mirroring port of a switch, or does it use the same NIC (and computer) as the softphone? You will first need to get in between the two endpoints in order to be able to capture that point-to-point RTP traffic - there are normal and malicious ways to achieve that. I have a switch that connects to the phone (10.1.0.38), PC with softphone (10.1.0.65), the Asterisk server and a VMHost that has the virtual machine where I use ettercap and tcpdump. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkvSRfcACgkQZpa/GxTmHTfCzQCdHhYG9ur6tuM+sd7q/v0on9RL pvAAnRw9coB7mtsF7PBFj0fQJ6mTw5Oo =3gN6 -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Security tests
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! In the network of my house I was testing the security with my Asterisk installation. The first test that I'm doing is an man in the middle attack. In this scenary, the attacker is a virtual machine that it tries to see the SIP traffic between a PC with a softphone and a Grandstream BT200 telephone. But it draws attention to me between the PC with softphone and the telephone I see traffic ARP or ICMP that could make to try between the equipment but does not see RTP. Is there some special consideration that it must to observe? I am doing it to the capture with: # tcpdump -i eth0 -n host 10.1.0.65 -w dump where 10.1.0.65 is the PC with softphone. Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkvPpYAACgkQZpa/GxTmHTenpwCfcL3gBTTf0jRiEpv0k+jf2GkP WR8An2RxSdFdkdyRntOmVUof5kOygLYB =EG9x -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote registering fails
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Alyed. On Sun, 11 Apr 2010, Alyed wrote: Daniel, you are having a problem often seen in pre 1.4.14 versions. Before this release srvlookup=no was the default for sip.conf and guess the same for iax.conf . So if you are working with a previous release just add this parameter .. but change it to serverlookup=yes under your iax.conf [general] section. Sorry, the parameter should be. srvlookup=yes I'm using Asterisk 1.4.24.1. Anyway, I was seeing the file sip.conf and yes I have srvlookup=yes in [general]. In iax.conf it is not defined explicitly, so I suppose that it will be taking the value by default. The context that I'm using for the local extensions is not [general]. Can it have to do? Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkvBw+sACgkQZpa/GxTmHTcdFQCfWiXsyRQ85s1fy9Ygb+IhlGGy 8kgAniMCjFLfZoyrEKKxao4FcRLsXTil =ltqS -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote registering fails
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I'm trying to test with a friend who has an Asterisk in his office with the Asterisk which I have in my house. Then I have an extension that he is trying to register remotely. Trying with the Twinkle client, I see that it is registered: - --- 400/400190.0.163.57 D N 5060 OK (35 ms) - --- but to the few seconds I obtain the following thing in Asterisk CLI: - --- 400/400190.0.163.57 D N 5060 UNREACHABLE - --- And Twinkle gives an error 408 request timeout. And when he tries to make the register through his Asterisk instead of use Twinkle, after a little while he obtains errors of this type: - --- [Apr 10 19:07:18] NOTICE[16848]: chan_sip.c:7618 sip_reg_timeout:-- Registration for '4...@myremotehome.com' timed out, trying again (Attempt #138) - --- This is the configuration that I'm using for the extension: - --- [400] username=400 type=friend secret=passwd qualify=yes callerid=Daniel 400 host=dynamic nat=no context=from-internal mailbox=...@voicemail canreinvite=no - --- I tried with both nat=yes ---as it is possible to be observed above--- and nat=no, and we always obtain the same behavior. My Asterisk server is installed in the same firewall with GNU/Linux. I don't believe that it is a problem with the ports since the client registers itself at some time. Whatever happens, I'm allowing connections for the remote IP to the 5060 tcp/UDP port and 1:2 UDP in the firewall. The router that it is ahead has these ports redirected to the firewall. Also I'm using externhost, externip and localnet in /etc/asterisk/sip.conf Which can be the problem? Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkvBAFQACgkQZpa/GxTmHTe0mgCcCmDNhkMm3DMc/Ckd7AAzZneF 4ngAn0SL/IC58kNDktcRsxJOaKPoAuCL =Ve4J -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Updating Asterisk and its use with MySQL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Alyed. On Sun, 28 Mar 2010, Alyed wrote: I didn't know that there was Digium's GUI. It is FLOSS? I was looking for in the site of Digium in the download section, but the unique thing that I saw that it speaks of a GUI is AsteriskNow, that in fact it is a complete distribution of GNU/Linux. You talked about to the GUI provided by AsteriskNow? Because if is this case, I don't believe that it is very practical. When I spoke of GUI was referring to a separated component to install over which already one had running. As far as the use of Asterisk with a DBMS (MySQL, for example), do you know some document or reference where indicate the steps to follow to migrate from config files? Yes I'm talking about Asterisk Now's GUI and yes, you can just install this component. google for Asterisk Gui 2.0 and you'll find plenty of info. Perfect. I will consider it. Thanks for the reference. In the tests that you said to me that you were doing, did you find this GUI as extensible as FreePBX? Regarding the DB I can't help you here, maybe someone else can. Well. If somebody can add something on this subject, will be welcome. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAku1HA0ACgkQZpa/GxTmHTdZnQCcDrmAPbVO6pCE1QF4YlBiftl9 v6oAn2YfB3s9RvqbqFt9/WSvX+TV4eqx =9PWK -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Updating Asterisk and its use with MySQL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Jim. On Sun, 28 Mar 2010, Jim Dickenson wrote: I think if you are installing dahdi complete from source you do make all and make install and make config Something that I forgot to ask previously is if the update of Asterisk or DAHDI is independent or the update of a component requires to also update the other. Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAku1IFAACgkQZpa/GxTmHTdZaACdGP8CAFLaGP2ek4pvdC2eHLOF 3noAnAhyWdDVboeGWzfP3Hw45s3jMPip =w4HG -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Updating Asterisk and its use with MySQL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I'm using Asterisk 1.4.24.1 with dahdi-linux-2.1.0.4 and dahdi-tools-2.1.0.2 compiled by myself with the source code of the official site of the project. I would like to update to one more newer version. I suppose that the recommendable thing is to maintain me in branch 1.4, reason why in this case it would be 1.4.30 that I suppose that it will have several bugs fixed. Also I see that there are new versions of DADHI Linux and DAHDI Tools; 2.2.1.1 for both cases. I image DAHDI Complete package include both DAHDI Linux an DAHDI tools. For this package it is necessary to continue making the compilation separately? But going to the question to that I make mention in subject, which would be the procedure to update the versions of these software maintaining the configurations? It is correct to think that the procedure would be to stop the Asterisk server and DAHDI, and to follow the same steps for the compilation and installation but without doing make config? On the other hand, at this moment I'm testing with few extensiones on low scale, but my idea is to raise the test a little more 50 extensions. For this case I suppose that it is more efficient to work with a database management system (MySQL, for example) for the configurations instead of files. There is some procedure that can recommend to me to migrate the configurations in files to a DBMS? My idea is to continue making the configurations by hand at the moment, that it is the way that I used until now, to familiarize to me with the handling of Asterisk at lower level, without using a graphical interface, and in a later stage of the tests to take these configurations through something like FreePBX. What think of this form to think? Thanks in advance for your reply and recommendations. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvcg0ACgkQZpa/GxTmHTdPNQCeL5oBGnuhcvqj8Sw8cuvUOBA8 DIoAn03AkmpGKN0XY1lMrLZ87RA2fhj4 =EKwq -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Updating Asterisk and its use with MySQL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Jim. On Sun, 28 Mar 2010, Jim Dickenson wrote: Make sure not to do make samples or you will overwrite your .conf file. This is the important one to watch out for. You can save off your .conf files and then restore them or compare your files with the new ones to see if there are any important new settings. I had thought that make config was what I would have to avoid. Which is the difference? does make config create the init scripts and make samples the example configuration files? Do these two makes have the same behavior for Asterisk and DAHDI? I have understood that make config in DAHDI Tools is the one that creates both the configuration files and init scripts. When I compiled the version that I'm using at the moment of DAHDI Linux only I used make and make install without using make samples or make config. Are also generated configuration files with DAHDI Linux? Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvjioACgkQZpa/GxTmHTcNIgCfZ1PEUqz/3kjGRTa0ECO97jSH 53YAni5ICLJGEL2U1Hcwc2hKsDUMYH6V =Dp0r -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Updating Asterisk and its use with MySQL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 - -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Jim. On Sun, 28 Mar 2010, Jim Dickenson wrote: Make sure not to do make samples or you will overwrite your .conf file. This is the important one to watch out for. You can save off your .conf files and then restore them or compare your files with the new ones to see if there are any important new settings. I had thought that make config was what I would have to avoid. Which is the difference? does make config create the init scripts and make samples the example configuration files? Yes, make config installs /etc/init.d/asterisk on Linux systems and does the appropriate chkconfig steps so will start on boot while make samples installs the .conf files in, by default, /etc/asterisk. Perfect. Do these two makes have the same behavior for Asterisk and DAHDI? I have understood that make config in DAHDI Tools is the one that creates both the configuration files and init scripts. There is no make config for dahdi. I think /etc/dahdi files do not get overwritten if they are there already. Hmmm... nevertheless I have documented this procedure in my Dokuwiki of the time that I made the installation and compilation: # tar xvzf dahdi-linux-2.1.0.4.tar.gz # tar xvzf dahdi-tools-2.1.0.2.tar.gz ~/Asterisk/dahdi-linux-2.1.0.4# make ~/Asterisk/dahdi-linux-2.1.0.4# make install ~/Asterisk/dahdi-tools-2.1.0.2# ./configure ~/Asterisk/dahdi-tools-2.1.0.2# make menuselect # In order to select a customized configuration ~/Asterisk/dahdi-tools-2.1.0.2# make ~/Asterisk/dahdi-tools-2.1.0.2# make install ~/Asterisk/dahdi-tools-2.1.0.2# make config # In order to install scripts and config files When I compiled the version that I'm using at the moment of DAHDI Linux only I used make and make install without using make samples or make config. Are also generated configuration files with DAHDI Linux? I think if you are installing dahdi complete from source you do make all and make install and make config Thanks. I will consider it if I install this package of DAHDI. Thanks for your reply. Regards, Daniel - -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvqhAACgkQZpa/GxTmHTfqMQCfT2V7RR4JMFp/EpH4J0F8Tfk9 3SYAoJJLhKfdznWoYddRNhmmyN1ygzJm =Q6s+ - -END PGP SIGNATURE- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvqh8ACgkQZpa/GxTmHTfoygCfZtRoPj8ieJjWVtsIqPFIk5Q/ 4QQAnjWRKkOJls9dFVwVM0IQORkmDIPd =YxoR -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Updating Asterisk and its use with MySQL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Sun, 28 Mar 2010, Alyed wrote: My idea is to continue making the configurations by hand at the moment, that it is the way that I used until now, to familiarize to me with the handling of Asterisk at lower level, without using a graphical interface, and in a later stage of the tests to take these configurations through something like FreePBX. What think of this form to think? I would suggest trying Digium's GUI first and then FreePBX since the first one I find it more readable. You'll find out eventually that there's no easy way to migrate from pure command line to a GUI, but you'll learn a lot in the meantime. I didn't know that there was Digium's GUI. It is FLOSS? I was looking for in the site of Digium in the download section, but the unique thing that I saw that it speaks of a GUI is AsteriskNow, that in fact it is a complete distribution of GNU/Linux. You talked about to the GUI provided by AsteriskNow? Because if is this case, I don't believe that it is very practical. When I spoke of GUI was referring to a separated component to install over which already one had running. As far as the use of Asterisk with a DBMS (MySQL, for example), do you know some document or reference where indicate the steps to follow to migrate from config files? Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvu8oACgkQZpa/GxTmHTdFVACePM0WaIfeHQmM+w8cpLuGGt/5 XSAAoI+YrC+9Y91ElRhFBrxAG6XVxEyh =e4iN -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail, Asterisk and Grandstream BT200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Alyed. On Mon, 22 Mar 2010, Alyed wrote: you are right, under [channels] is where it's supposed to be my mistake, i guess i was thinking in sip.conf :) Perfect :-) However, the following doubt arises to me: it would also have had this problem for some originating call from a telephone that is not a cell phone? yes, and this can be a really serious problem if you don't fix it. So don't forget to include this parameters from now on. I have played with them and found setting busycount=5 is not very efficent, so leave it to 3 or 4 at most. That problematic. I will consider it in future configurations. Thanks for the explanation. Good to hear your problem is solved. Thanks again for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkurK40ACgkQZpa/GxTmHTc/7gCfWbbEsqrXrWkM8ByGhQKLun1o kJ0AnA0mB+5+Rn9sWZ20iVemMXZJfhX6 =Vytb -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail, Asterisk and Grandstream BT200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Alyed. On Mon, 22 Mar 2010, Alyed wrote: I was with the following situation: if I call from a cell phone, my Asterisk take the call, it presents to the caller the possibility to dialing an extension number and, in case of not doing it, it transfers this call to a specific extension. Then, if in this extension nobody takes the call, the service of voicemail is triggered so that the caller leaves its message from the cell phone. But if it hangs after to let the message without have pressed previously the pound key, the channel is taken and no longer any other call enters the PBX from the PSTN. This does not happen if the caller presses the pound key after to have left his message. As I have a box at which the cable arrives from the PSTN in which there are two ports of derivation and in one of them it leaves the cable for the Asterisk PBX (connected only then), after to have detected this problem I tried connecting in the other port an analog telephone and, indeed, it did not have tone as if never it had been hung. In addition this was confirmed because the MWI light never blinked on the telephone. After restarting the Asterisk server, yes the MWI light blinks and in addition I could corob the time in which the channel was taken seeing that the message lasted more than nine minutes. To what this problem can be due? It has to do the call is made specifically from cell phone through the PSTN (because if I leave a message hanging directly without pressing the pound key from an local extension, this does not happen)? There is some form to avoid it? Make sure you have busydetect=yes busycount=3 somewhere below your [general] context in chan_dahdi.conf (or zapata.conf depending on your asterisk version) and restart the the service. This should be enoough to do the magic. It didn't have configured these two parameters so I added now them but in the [channels] context since I don't have a [general] context (It does not sound to me that in the file by default generated by Asterisk there would not be it either, although I can be mistaken). Beyond that, with these two parameters, I no longer have the problem mentioned before. Thanks! However, the following doubt arises to me: it would also have had this problem for some originating call from a telephone that is not a cell phone? Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkunNjQACgkQZpa/GxTmHTfAbACfT8PVkcp/xESdqsiczg3YY/Dd FGcAn1TdOqiZaKAjLg4h3SDt/34A4bKX =37qZ -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail, Asterisk and Grandstream BT200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Gordon. On Sun, 21 Mar 2010, Gordon Henderson wrote: I'm testing with a Grandstream BT200 telephone and, according to I read, it has a LED that blinks if for that extension messages were left. In Voice Mail UserID, under the ACCOUNT tab, I put *100 that is the extension in which my Asterisk answer the voicemail service and if then I press MESSAGE button, the telephone communicates with Asterisk and, after to introduce the password, it indicates to me that I have messages. But the luminous indicator does not work. It is necessary to configure something special for this? It can be that it doesn't work because there is to introduce one password previously? There's another setting in the phone you need to set SUBSCRIBE for MWI. Yes. I was needing to indicate the use of MWI of the side of the configuration of the telephone. I selected the SUBSCRIBES for MWI checkbox. And make-sure the mailbox number is listed in the sip.conf entry for that phone. According to which I was reading, the MWI notifications become by the option mailbox= in the configuration of the extension. For this extension, the 104, had mailbox=104 but still with MWI enabled option, it was not working. After to think enough on this subject, I have noticed that instead of 104 I had to put 1...@voicemail since voicemail it was context that I'm using in voicemail.conf. With this already was working. However, beyond this, I was with the following situation: if I call from a cell phone, my Asterisk take the call, it presents to the caller the possibility to dialing an extension number and, in case of not doing it, it transfers this call to a specific extension. Then, if in this extension nobody takes the call, the service of voicemail is triggered so that the caller leaves its message from the cell phone. But if it hangs after to let the message without have pressed previously the pound key, the channel is taken and no longer any other call enters the PBX from the PSTN. This does not happen if the caller presses the pound key after to have left his message. As I have a box at which the cable arrives from the PSTN in which there are two ports of derivation and in one of them it leaves the cable for the Asterisk PBX (connected only then), after to have detected this problem I tried connecting in the other port an analog telephone and, indeed, it did not have tone as if never it had been hung. In addition this was confirmed because the MWI light never blinked on the telephone. After restarting the Asterisk server, yes the MWI light blinks and in addition I could corob the time in which the channel was taken seeing that the message lasted more than nine minutes. To what this problem can be due? It has to do the call is made specifically from cell phone through the PSTN (because if I leave a message hanging directly without pressing the pound key from an local extension, this does not happen)? There is some form to avoid it? Thanks for your reply! Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkums0oACgkQZpa/GxTmHTcGpQCghJvfphxc5ZzZhouryA+OlwGm 20AAoJP64a2EVeigx08D/5g5XN8oBXgf =Hskd -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail, Asterisk and Grandstream BT200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I'm testing with a Grandstream BT200 telephone and, according to I read, it has a LED that blinks if for that extension messages were left. In Voice Mail UserID, under the ACCOUNT tab, I put *100 that is the extension in which my Asterisk answer the voicemail service and if then I press MESSAGE button, the telephone communicates with Asterisk and, after to introduce the password, it indicates to me that I have messages. But the luminous indicator does not work. It is necessary to configure something special for this? It can be that it doesn't work because there is to introduce one password previously? Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkulXlgACgkQZpa/GxTmHTfQBACfUkqoST6HRgpsXwcBZpXfLdan UaoAn2peX4pmoe3GlgoBL9GcOBxmg9UR =0RBM -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registering of Asterisk against a SIP provider
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday, Feb 18, 2010 at 05:36:41 -0300, Administrator TOOTAI wrote: Hi Hi, Daniel. Daniel Bareiro a écrit : [...] Hours ago the IP changed and the domain was updated satisfactorily, but in spite of this I was obtaining the registering failures that I mentioned above. After to restart Asterisk (1.4.24.1), I no longer had this problem of registering. But there would be some way to solve this problem? [...] It's an old story. Asterisk check DNS when it start that's why it's ok after you have it restarted. When I was running Asterisk using dynamic addresses, I made following: - modify sip.conf to include a file placed where ever you want, contents being externalip/externalhosts and all others info needed related to external IP - restarted myself ADSL line with a cron script each night - this script extract/found the new IP using the method you prefer (eg ping your dyndns host until response and than you have your new IP and insert the IP in the file you include in sip.conf - this script restart asterisk and voila :-) Was working like a charm. As I said to Warren, according to the tests that I was doing, apparently this can be solved with both externip and externhost,restarting Asterisk in either cases. In the case of externhost we would be saving ourselves to have to modify the IP in sip.conf every time, but even so we would have to verify if the IP has changed for restarting Asterisk. I thought that perhaps this could be solved without restarting Asterisk. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkt+kkYACgkQZpa/GxTmHTcpxwCfbwAbaYEzEv6rBqZIWQs5kLER STkAn00FEXGzD+berHCZYe20HLBnXZQU =xzIk -END PGP SIGNATURE- -- _ -- 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] Error redirecting an incoming call of a SIP provider to a local extension
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I am trying to redirect to a local extension the incoming calls that I
receive to an account which I have in iptel.org, but when receiving I'm
obtaining this error:
alderamin*CLI
-- Executing [...@from-internal:1] Dial(SIP/danib-089f8820,
SIP/300|30|tTrm) in new stack
[Feb 19 19:22:50] WARNING[19254]: app_dial.c:1237 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
[Feb 19 19:23:00] WARNING[19254]: pbx.c:2529 __ast_pbx_run: Timeout, but
no rule 't' in context 'from-internal'
It is probable that this can be due to a problem of interaction between
contexts? I copy the content of extensions.conf and sip.conf to see if
it can help to find the problem:
-
extensions.conf:
; DGB - 20091114
[general]
autofallthrough=no
[macro-dial]
exten = s,1,Dial(${ARG1},15)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${macro_ext...@voicemail,u)
exten = s-NOANSWER,n,Hangup
exten = s-BUSY,1,Voicemail(${macro_ext...@voicemail,b)
exten = s-BUSY,n,Hangup
exten = s-CHANUNAVAIL,1,Playback(pbx-invalid)
[from-internal]
; Llamadas a extensiones SIP
exten = _2xx,1,Macro(dial,SIP/${EXTEN})
exten = _2xx,n,Hangup
exten = 300,1,Dial(SIP/300,30,tTrm)
; Extension analogica
exten = 402,1,Macro(dial,DAHDI/2)
exten = 402,n,Hangup
; Directorio de extensiones
exten = *400,1,Directory(voicemail,from-internal)
; Musica en espera
exten = *300,1,Answer
exten = *300,n,SetMusicOnHold(default)
exten = *300,n,WaitMusicOnHold(2000)
exten = *300,n,Hangup
; Prueba de Eco
exten = *200,1,Answer
exten = *200,n,Playback(demo-echotest)
exten = *200,n,Echo
exten = *200,n,Playback(demo-echodone)
exten = *200,n,Hangup
; Acceso a voicemail
exten = *100,1,Answer
exten = *100,n,Wait(1)
exten = *100,n,VoiceMailMain(${CALLERID(num)}...@voicemail)
exten = *100,n,Hangup
; Llamadas salientes
exten = _9.,1,Dial(DAHDI/1/${EXTEN:1})
exten = _9.,n,Hangup
; Call a number at iptel.org
exten = _0.,1,Dial(SIP/iptel/${EXTEN:1},20,r))
exten = _0.,n,Hangup
[from-pstn]
; incoming calls from FXO port are directed to this context
exten = s,1,Answer()
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=15)
exten = s,n,Background(contestador1)
exten = i,1,Goto(from-pstn,s,1)
exten = t,1,Playback(locomunicoconelinterno1)
exten = t,n,Dial(SIP/200,25)
exten = t,n,VoiceMail(2...@voicemail,20)
exten = t,n,Hangup()
include = from-internal
-
sip.conf:
[general]
[...]
; register with iptel.org
register = danib:mlrzv...@iptel.org/300
[...]
; Outgoing to iptel.org
[iptel]
type=friend
username=danib
secret=myspasswd
host=iptel.org
canreinvite=no
qualify=300
insecure=port,invite ; required for incoming ekiga.net calls
context = from-internal
-
Thanks in advance for your replies.
Regards,
Daniel
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Version: GnuPG v1.4.9 (GNU/Linux)
iEYEARECAAYFAkt/LkUACgkQZpa/GxTmHTeglwCgh8E59wZ+9yBXEWhwC+RdnZgP
16MAnRh4NDaN9QOGHjIRbvWUQtiA2v23
=6iU8
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Re: [asterisk-users] Registering of Asterisk against a SIP provider
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Warren. On Thursday, Feb 18, 2010 at 00:01:23 -0300, Warren Selby wrote: ; DGB - 20100211 externip = sysadminhaiku.com.ar localnet = 10.1.0.0/24 If you're using dynamic dns, shouldn't you be using externhost instead of externip? It can be. I was using externip because I found this reference in the Web on the recommendation to use it to somebody having registering problems. But I'm going to test this that you mention to me. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkt9lWkACgkQZpa/GxTmHTe5IACfYNjZlblhxaX9tVhWeD8KeTJt //gAnivaaIcpE3a8+Nw5G0sW5cmUduZe =hBTi -END PGP SIGNATURE- -- _ -- 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] Registering of Asterisk against a SIP provider
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Warren. On Thursday, Feb 18, 2010 at 16:30:40 -0300, Daniel Bareiro wrote: ; DGB - 20100211 externip = sysadminhaiku.com.ar localnet = 10.1.0.0/24 If you're using dynamic dns, shouldn't you be using externhost instead of externip? It can be. I was using externip because I found this reference in the Web on the recommendation to use it to somebody having registering problems. But I'm going to test this that you mention to me. Changing the line of externip by the following one: externhost = sysadminhaiku.com.ar and forcing the router to change the public IP, I'm observing the same message that before I've commented even after to have restarted Asterisk: [Feb 18 17:24:50] NOTICE[20328]: chan_sip.c:7715 sip_reg_timeout:-- Registration for 'dan...@ekiga.net' timed out, trying again (Attempt #17) -- Got SIP response 606 Not Acceptable back from 86.64.162.35 Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkt9o0AACgkQZpa/GxTmHTeNAwCeOEd3G2JcpH9YN1urIoM5gI9E rLcAn2Z+SiRtMt94rQdXuGSe6MRW0eMd =Jyag -END PGP SIGNATURE- -- _ -- 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] Registering of Asterisk against a SIP provider
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday, Feb 18, 2010 at 17:29:44 -0300, Daniel Bareiro wrote: ; DGB - 20100211 externip = sysadminhaiku.com.ar localnet = 10.1.0.0/24 If you're using dynamic dns, shouldn't you be using externhost instead of externip? It can be. I was using externip because I found this reference in the Web on the recommendation to use it to somebody having registering problems. But I'm going to test this that you mention to me. Changing the line of externip by the following one: externhost = sysadminhaiku.com.ar and forcing the router to change the public IP, I'm observing the same message that before I've commented even after to have restarted Asterisk Correction: if I restart Asterisk, the registering fails six times but after that it is registered. Then, it as much seems that with both externip and externhost, a restart of Asterisk is required. I thought that perhaps there would be some way to avoid this... Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkt90DMACgkQZpa/GxTmHTfVSgCghupu+0PhJp159q7MglNjiZ6s //kAn1NaY29MaOLe55BtPgKN+td9KN5M =rGty -END PGP SIGNATURE- -- _ -- 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] Registering of Asterisk against a SIP provider
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, all! I'm being based on this document [1] to send and to receive calls using ekiga.net. But I'm seeing, in an Asterisk console, several messages of this type: [Feb 17 21:19:15] NOTICE[11875]: chan_sip.c:7715 sip_reg_timeout:-- Registration for 'dan...@ekiga.net' timed out, trying again (Attempt #4775) -- Got SIP response 606 Not Acceptable back from 86.64.162.35 Investigating in Internet I found that it can be due to that the registering is being tried to do with an not public IP. I've dynamic IP whose domain is updated using a dynamic DNS service. The line that I am using in sip.conf is the following one: ; DGB - 20100211 externip = sysadminhaiku.com.ar localnet = 10.1.0.0/24 Hours ago the IP changed and the domain was updated satisfactorily, but in spite of this I was obtaining the registering failures that I mentioned above. After to restart Asterisk (1.4.24.1), I no longer had this problem of registering. But there would be some way to solve this problem? Thanks in advance for your replies. Regards, Daniel [1] http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkt8j5oACgkQZpa/GxTmHTcV3QCeIyaEg9/1KtT7tOnDTGVLKhXl gvkAn1Cj8c52a30Hf1GX4bWnOyALbwpX =Vl61 -END PGP SIGNATURE- -- _ -- 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] Recommendations about infrastructure to use with Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I'm investigating the possibility of using Asterisk as much for internal communication in an office as between offices and I would like to know what considerations you could comment to me being based on the experience that you have had. A priori two things come to my mind: * As to network topology, is advisable to have switches and dedicated networks for to use with the extensions? * Is advisable to have a dedicated Internet connection for intercommunication between the different offices? I imagine that yes, since of another way the VoIP traffic would have to compete with the rest and in that case we would require to apply some additional technique of QoS. In this point also I would include the optimal bandwidth that would have to have the dedicated link, for the case of using something of this type. Perhaps there is some other interesting questions that also it is necessary to consider. In order to give more additional information, the Internet connection between the different offices is made at the moment through two links of 2 Mbps, with load balance (one of fiber and another one of microwaves). The amount of extensions in one of the offices would be approximately of 50, whereas in the other there would be approximately about 80 extensions. Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkqfjPAACgkQZpa/GxTmHTfm8ACfXUHf8helAFxo5Tqmjk6TCiq2 5CwAnAyfGsCVEL+6g7O2juTPnLh9gHIj =v8+9 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Accessing to ekiga.net through Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 El miércoles 19 de agosto del 2009 a las 08:04:17 -0300, SIP escribió: Daniel, Hi SIP. I'm a little confused as to what I'm seeing here. You're bounding through two RFC1918 address networks -- 10.1.0.X and 192.168.2.X. Is this some sort of dual NAT scenario? Perhaps if you can explain a little more about your network setup. This it is a scheme of my network configuration: +--+ +-+ ___/\__ | | | | / \ | GNU/Linux eth1-+ ADSL Router +-| Internet | | Firewall/ | | | \__ __/ | Asterisx eth0++-+ \_/ | | | +--+ | | +--+--+ | LAN switch | +-+ The ADSL router is configured to connect itself to Internet for its own means (I don't use any software PPPoE in the GNU/Linux box). This router uses the private IP 192.168.1.1. In the GNU/Linux box the eth1 interface uses the private IP 192.168.1.2. The eth0 interface (10.1.0.10) is the point of connection to the rest of the LAN (10.1.0.0/24). Firewall makes NAT of all the originating traffic of eth0 through eth1. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkqZQ/8ACgkQZpa/GxTmHTcxAgCfQwO4PxNarZO7nAFwQSVK1EW/ /wYAnR3KQF6+6p2jkKo1spZxi1RjT4de =gCuK -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Accessing to ekiga.net through Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 SIP wrote: Daniel, Hi SIP. Check your stunaddr setting. Is it misspelled, or do they really use stun.exiga.net instead of stun.ekiga.net ? Thanks to indicate that error to me. I doing the test again. I don't believe that this solves what I commented before about 192.168.1.2 direction, but, just in case, I copy the output of debugging when trying to communicate to ekiga.net. The problem continues persisting after the correction. alderamin*CLI --- SIP read from 10.1.0.65:5060 --- INVITE sip:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke Max-Forwards: 70 To: sip:8...@10.1.0.10 From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 709 INVITE Contact: sip:2...@10.1.0.65 Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2 Content-Length: 247 v=0 o=twinkle 933572867 1938524932 IN IP4 10.1.0.65 s=- c=IN IP4 10.1.0.65 t=0 0 m=audio 8000 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 - - --- (13 headers 12 lines) --- Sending to 10.1.0.65 : 5060 (NAT) Using INVITE request as basis request - kafgeaflkmsd...@defiant.freesoftware.org --- Reliably Transmitting (no NAT) to 10.1.0.65:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.0.65;branch=z9hG4bKrqslryke;received=10.1.0.65;rport=5060 From: Hector sip:2...@10.1.0.10;tag=typwm To: sip:8...@10.1.0.10;tag=as0a3a462b Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 709 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=497d879d Content-Length: 0 Scheduling destruction of SIP dialog 'kafgeaflkmsd...@defiant.freesoftware.org' in 32000 ms (Method: INVITE) Found user '201' alderamin*CLI --- SIP read from 10.1.0.65:5060 --- ACK sip:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke Max-Forwards: 70 To: sip:8...@10.1.0.10;tag=as0a3a462b From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 709 ACK User-Agent: Twinkle/1.2 Content-Length: 0 - - --- (9 headers 0 lines) --- alderamin*CLI --- SIP read from 10.1.0.65:5060 --- INVITE sip:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKxpraybjr Max-Forwards: 70 Proxy-Authorization: Digest username=201,realm=asterisk,nonce=497d879d,uri=sip:8...@10.1.0.10,response=9cb53107d4d15b7a2e7df8599e851b80,algorithm=MD5 To: sip:8...@10.1.0.10 From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 710 INVITE Contact: sip:2...@10.1.0.65 Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2 Content-Length: 247 v=0 o=twinkle 933572867 1938524932 IN IP4 10.1.0.65 s=- c=IN IP4 10.1.0.65 t=0 0 m=audio 8000 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 - - --- (14 headers 12 lines) --- Sending to 10.1.0.65 : 5060 (NAT) Using INVITE request as basis request - kafgeaflkmsd...@defiant.freesoftware.org Found user '201' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 10.1.0.65:8000 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4040a (gsm|alaw|ilbc|h261), peer - audio=0xe (gsm|ulaw| alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.1.0.65:8000 Looking for 8500 in from-internal (domain 10.1.0.10) list_route: hop: sip:2...@10.1.0.65 --- Transmitting (no NAT) to 10.1.0.65:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.65;branch=z9hG4bKxpraybjr;received=10.1.0.65;rport=5060 From: Hector sip:2...@10.1.0.10;tag=typwm To: sip:8...@10.1.0.10 Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 710 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:8...@10.1.0.10 Content-Length: 0 -- Executing [8...@from-internal:1] Dial(SIP/201-0900, SIP/ekiga/500|20|r)) in new stack Video is at 192.168.1.2 port 10112 Audio is at 192.168.1.2 port 12592 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (h261) to SDP Adding non-codec 0x1 (telephone-event)
[asterisk-users] Accessing to ekiga.net through Asterisk
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I'm trying to connect to ekiga.net through a client connected to my
Asterisk server. For it I am being based on this [1] document. Next I
put the configurations that I am using.
/etc/asterisk/sip.conf:
; Outgoing to ekiga.net
[ekiga]
type=friend
username=MyUser
secret=MyPass
host=ekiga.net
canreinvite=no
qualify=300
nat = yes
stunaddr = stun.exiga.net
insecure=port,invite ; required for incoming ekiga.net calls
/etc/asterisk/extensions.conf:
[from-internal]
...
exten = _8.,1,Dial(SIP/ekiga/${EXTEN:1},20,r))
I tried a echo test, dialing in my case to 8500, but in spite of seeing
traffic towards Internet, nothing is heard. Setting 'sip set debug', I get
the following thing:
--- SIP read from 10.1.0.65:5060 ---
INVITE sip:8...@10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
Max-Forwards: 70
To: sip:8...@10.1.0.10
From: Hector sip:2...@10.1.0.10;tag=uucwz
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq: 183 INVITE
Contact: sip:2...@10.1.0.65
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.2
Content-Length: 247
v=0
o=twinkle 2122879389 441437466 IN IP4 10.1.0.65
s=-
c=IN IP4 10.1.0.65
t=0 0
m=audio 8000 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
-
- --- (13 headers 12 lines) ---
Sending to 10.1.0.65 : 5060 (NAT)
Using INVITE request as basis request - mrsyiysrdkwm...@defiant.freesoftware.org
--- Reliably Transmitting (no NAT) to 10.1.0.65:5060 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.1.0.65;branch=z9hG4bKjxcxrrks;received=10.1.0.65;rport=5060
From: Hector sip:2...@10.1.0.10;tag=uucwz
To: sip:8...@10.1.0.10;tag=as095989a3
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq: 183 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=76b2dfe8
Content-Length: 0
Scheduling destruction of SIP dialog 'mrsyiysrdkwm...@defiant.freesoftware.org'
in 32000 ms (Method: INVITE)
Found user '201'
alderamin*CLI
--- SIP read from 10.1.0.65:5060 ---
ACK sip:8...@10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
Max-Forwards: 70
To: sip:8...@10.1.0.10;tag=as095989a3
From: Hector sip:2...@10.1.0.10;tag=uucwz
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq: 183 ACK
User-Agent: Twinkle/1.2
Content-Length: 0
-
- --- (9 headers 0 lines) ---
alderamin*CLI
--- SIP read from 10.1.0.65:5060 ---
INVITE sip:8...@10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKoilauqhp
Max-Forwards: 70
Proxy-Authorization: Digest
username=201,realm=asterisk,nonce=76b2dfe8,uri=sip:8...@10.1.0.10,response=d49c0fdf11c9977fcd1fce6a50f445fe,algorithm=MD5
To: sip:8...@10.1.0.10
From: Hector sip:2...@10.1.0.10;tag=uucwz
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq: 184 INVITE
Contact: sip:2...@10.1.0.65
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.2
Content-Length: 247
v=0
o=twinkle 2122879389 441437466 IN IP4 10.1.0.65
s=-
c=IN IP4 10.1.0.65
t=0 0
m=audio 8000 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
-
- --- (14 headers 12 lines) ---
Sending to 10.1.0.65 : 5060 (NAT)
Using INVITE request as basis request - mrsyiysrdkwm...@defiant.freesoftware.org
Found user '201'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 10.1.0.65:8000
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4040a (gsm|alaw|ilbc|h261), peer - audio=0xe
(gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.0.65:8000
Looking for 8500 in from-internal (domain 10.1.0.10)
list_route: hop: sip:2...@10.1.0.65
--- Transmitting (no NAT) to 10.1.0.65:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.0.65;branch=z9hG4bKoilauqhp;received=10.1.0.65;rport=5060
From: Hector sip:2...@10.1.0.10;tag=uucwz
To: sip:8...@10.1.0.10
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq: 184 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:8...@10.1.0.10
Content-Length: 0
-- Executing [8...@from-internal:1]
Re: [asterisk-users] Accessing to ekiga.net through Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 SIP wrote: Daniel, Hi SIP. Check your stunaddr setting. Is it misspelled, or do they really use stun.exiga.net instead of stun.ekiga.net ? Thanks to indicate that error to me. I doing the test again. I don't believe that this solves what I commented before about 192.168.1.2 direction, but, just in case, I copy the output of debugging when trying to communicate to ekiga.net. alderamin*CLI --- SIP read from 10.1.0.65:5060 --- INVITE sip:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke Max-Forwards: 70 To: sip:8...@10.1.0.10 From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 709 INVITE Contact: sip:2...@10.1.0.65 Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2 Content-Length: 247 v=0 o=twinkle 933572867 1938524932 IN IP4 10.1.0.65 s=- c=IN IP4 10.1.0.65 t=0 0 m=audio 8000 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 - - --- (13 headers 12 lines) --- Sending to 10.1.0.65 : 5060 (NAT) Using INVITE request as basis request - kafgeaflkmsd...@defiant.freesoftware.org --- Reliably Transmitting (no NAT) to 10.1.0.65:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.0.65;branch=z9hG4bKrqslryke;received=10.1.0.65;rport=5060 From: Hector sip:2...@10.1.0.10;tag=typwm To: sip:8...@10.1.0.10;tag=as0a3a462b Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 709 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=497d879d Content-Length: 0 Scheduling destruction of SIP dialog 'kafgeaflkmsd...@defiant.freesoftware.org' in 32000 ms (Method: INVITE) Found user '201' alderamin*CLI --- SIP read from 10.1.0.65:5060 --- ACK sip:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke Max-Forwards: 70 To: sip:8...@10.1.0.10;tag=as0a3a462b From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 709 ACK User-Agent: Twinkle/1.2 Content-Length: 0 - - --- (9 headers 0 lines) --- alderamin*CLI --- SIP read from 10.1.0.65:5060 --- INVITE sip:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKxpraybjr Max-Forwards: 70 Proxy-Authorization: Digest username=201,realm=asterisk,nonce=497d879d,uri=sip:8...@10.1.0.10,response=9cb53107d4d15b7a2e7df8599e851b80,algorithm=MD5 To: sip:8...@10.1.0.10 From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 710 INVITE Contact: sip:2...@10.1.0.65 Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2 Content-Length: 247 v=0 o=twinkle 933572867 1938524932 IN IP4 10.1.0.65 s=- c=IN IP4 10.1.0.65 t=0 0 m=audio 8000 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 - - --- (14 headers 12 lines) --- Sending to 10.1.0.65 : 5060 (NAT) Using INVITE request as basis request - kafgeaflkmsd...@defiant.freesoftware.org Found user '201' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 10.1.0.65:8000 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4040a (gsm|alaw|ilbc|h261), peer - audio=0xe (gsm|ulaw| alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.1.0.65:8000 Looking for 8500 in from-internal (domain 10.1.0.10) list_route: hop: sip:2...@10.1.0.65 --- Transmitting (no NAT) to 10.1.0.65:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.65;branch=z9hG4bKxpraybjr;received=10.1.0.65;rport=5060 From: Hector sip:2...@10.1.0.10;tag=typwm To: sip:8...@10.1.0.10 Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 710 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:8...@10.1.0.10 Content-Length: 0 -- Executing [8...@from-internal:1] Dial(SIP/201-0900, SIP/ekiga/500|20|r)) in new stack Video is at 192.168.1.2 port 10112 Audio is at 192.168.1.2 port 12592 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (h261) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to
[asterisk-users] Recommendation / doubt about building of dialplan
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
Now that I have a little more time, I was debugging my dialplan and it
was of the following way:
- -
; DGB - 20090615
[macro-dial]
exten = s,1,Dial(${ARG1},15)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${macro_ext...@voicemail,u)
exten = s-NOANSWER,n,Hangup
exten = s-BUSY,1,Voicemail(${macro_ext...@voicemail,b)
exten = s-BUSY,n,Hangup
exten = s-CHANUNAVAIL,1,Playback(pbx-invalid)
[from-internal]
; Call to SIP extensions
exten = _xxx,1,Macro(dial,SIP/${EXTEN})
exten = _xxx,n,Hangup
; Analog extension
exten = 402,1,Macro(dial,DAHDI/2)
exten = 402,n,Hangup
; Outgoing calls
exten = _9.,1,Dial(DAHDI/1/${EXTEN:1})
exten = _9.,n,Hangup
;exten = 9,1,Dial(DAHDI/1,20,tTr)
; Voicemail
exten = *100,1,Answer
exten = *100,n,Wait(1)
exten = *100,n,VoiceMailMain(${CALLERID(num)}...@voicemail)
exten = *100,n,Hangup
; Echo test
exten = *200,1,Answer
exten = *200,n,Playback(demo-echotest)
exten = *200,n,Echo
exten = *200,n,Playback(demo-echodone)
exten = *200,n,Hangup
; Music on the hold
exten = *300,1,Answer
exten = *300,n,SetMusicOnHold(default)
exten = *300,n,WaitMusicOnHold(2000)
exten = *300,n,Hangup
; Dial-by-name directory
exten = *400,1,Directory(voicemail,from-internal)
;---
[from-pstn]
; incoming calls from FXO port are directed to this context
exten = s,1,Dial(DAHDI/2,15,tTrm)
exten = s,n,Background(if-u-know-ext-dial) ; Dial known extension
exten = s,n,WaitExten()
include = from-internal
- -
Although internally it works as I had thought in such a way that
Asterisk derives to the voicemail indicating the reason by which one
became (nonavailable person or busy extension) and to indicate that the
extension is not valid in case it does not exist or the extension is not
registered when to try to contact (if there is some situation that I'm
ignoring, make to me notice it, please), the problem that I am seeing
with this is that if I include from-iternal context in from-pstn in such
a way that the incoming calls from the PSTN can communicate with both
SIP or DAHDI extensions, I think (with my present knowledge of Asterisk)
that it will be not useful to me so that in case the extension is not
valid a Goto(s,2) of from-pstn are accomplished so that the person can
dial the extension again without having to make a new call.
I suppose that it would be possible to be done defining again the
extensions in context from-pstn, but I suppose that there will be one
more efficient way to obtain the behavior to which I made reference of
one better way, which can be especially useful if we have defined a lot
of extensions.
Thanks in advance for your reply.
Regards,
Daniel
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Re: [asterisk-users] Transfer call from analog telephone
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Daniel Bareiro wrote:
As I've commented in a previous message, after dial *60 (of *600 to Echo
test), I obtain like a tone cut in three parts followed of a continuous tone,
causing that I'm incapable to dial the extension completely. The
waitfordigit appears after to hangup. The cell_number seems to be some
number that I has dial previously. Testing again with a SIP extension, this
problem does not happen.
Also it draws attention to me that the DTMF has a duration of 0ms.
It is peculiar... after to have a restart of Asterisk, I can dial without
problems to *600. This is Asterisk log corresponding to the successful
communication with the extension:
- --
[Jun 4 23:03:30] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '*'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:03:30] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted
without begin '*' on DAHDI/2-1
[Jun 4 23:03:30] DTMF[28905]: channel.c:2282 __ast_read: DTMF end
passthrough '*' on DAHDI/2-1
[Jun 4 23:03:30] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '6'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:03:30] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted
without begin '6' on DAHDI/2-1
[Jun 4 23:03:30] DTMF[28905]: channel.c:2282 __ast_read: DTMF end
passthrough '6' on DAHDI/2-1
[Jun 4 23:03:31] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '0'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:03:31] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted
without begin '0' on DAHDI/2-1
[Jun 4 23:03:31] DTMF[28905]: channel.c:2282 __ast_read: DTMF end
passthrough '0' on DAHDI/2-1
[Jun 4 23:03:31] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '0'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:03:31] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted
without begin '0' on DAHDI/2-1
[Jun 4 23:03:31] DTMF[28905]: channel.c:2282 __ast_read: DTMF end
passthrough '0' on DAHDI/2-1
-- Executing [*...@phones:1] Answer(DAHDI/2-1, ) in new stack
[Jun 4 23:03:31] DEBUG[28905]: chan_dahdi.c:3174 dahdi_answer: Took
DAHDI/2-1 off hook
-- Executing [*...@phones:2] Playback(DAHDI/2-1, demo-echotest) in
new stack
-- DAHDI/2-1Playing 'demo-echotest' (language 'es')
== Spawn extension (phones, *600, 2) exited non-zero on 'DAHDI/2-1'
-- Hungup 'DAHDI/2-1'
- --
As you will see, the duration is always of 0 ms (also when I dial to the cell
phone). After this I make several tests. To dial from cell phone to the analog
phone and I did not have problems in to call immediately to *600 after to have
dial to the cell phone in each opportunity. But if from my extension 201 I
dial the analog phone and after that from my analog phone I dial to *600, it
happens the same of problem of not to be able to dial beyond *60. Log of the
CLI for this situation is the following one:
- --
[Jun 4 23:08:45] DTMF[29017]: channel.c:2229 __ast_read: DTMF end '*'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:08:45] DTMF[29017]: channel.c:2271 __ast_read: DTMF end accepted
without begin '*' on DAHDI/2-1
[Jun 4 23:08:45] DTMF[29017]: channel.c:2282 __ast_read: DTMF end
passthrough '*' on DAHDI/2-1
[Jun 4 23:08:46] DTMF[29017]: channel.c:2229 __ast_read: DTMF end '6'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:08:46] DTMF[29017]: channel.c:2271 __ast_read: DTMF end accepted
without begin '6' on DAHDI/2-1
[Jun 4 23:08:46] DTMF[29017]: channel.c:2282 __ast_read: DTMF end
passthrough '6' on DAHDI/2-1
[Jun 4 23:08:46] DTMF[29017]: channel.c:2229 __ast_read: DTMF end '0'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:08:46] DTMF[29017]: channel.c:2271 __ast_read: DTMF end accepted
without begin '0' on DAHDI/2-1
[Jun 4 23:08:46] DTMF[29017]: channel.c:2282 __ast_read: DTMF end
passthrough '0' on DAHDI/2-1
-- Blacklisting number 201
[Jun 4 23:08:54] DEBUG[29017]: chan_dahdi.c:6244 ss_thread: waitfordigit
returned 0...
-- Hungup 'DAHDI/2-1'
- --
Testing some more I could verify than if I changed the number for echo test to
*700 instead of *600, the problem of not being able to dial beyond *60
disappears. Investigating a little in Internet and reading the source code, I
found the following in the line 2834 of chan_mgcp.c file:
- -
2834 } else if (!ast_strlen_zero(p-lastcallerid) !strcmp(p-dtmf_buf,
*60)) {
2835 if (option_verbose 2) {
2836 ast_verbose(VERBOSE_PREFIX_3 Blacklisting number
%s\n, p-lastcallerid);
2837 }
2838 res = ast_db_put
Re: [asterisk-users] Transfer call from analog telephone
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Tilghman and Grygoriy. Tilghman Lesher escribió: I was testing both the recall key and uncomment the following lines in the features.conf file: blindxfer = #1 atxfer = *2 verifying previously that the extension uses the arguments tT with the Dial application and to include the context featuremap in the context in which I have defined the extensions (internal). The telephone of the end with which the conversation is staying listens the tones to try doing the transfer, but Asterisk does not give the dial tone after *2 / #1 or the recall key. Remember that the time between the two digits is VERY short. You must press those two digits in quick succession or else the requested feature code will not activate. I made sure to make it sufficiently fast, but still increasing featuredigittimeout, it did not work. I am not sure if it will have some relation, but also found another difficulty when the dial from my analog telephone. When doing a echo test from an SIP extension, I don't have problems, but, sometimes, with an analog telephone when trying to dial the extension to realise the echo test (*600), after to have dial *60, a tone cut in three parts is listened to soon a continuous tone, doing impossible to be able to dial the extension completely. Sometimes it works well, but sometimes it happens, that is something that draws attention to me and, as it mentioned, from a SIP extension I'm not having this problem. This is what I get in the Asterisk CLI after to dial *60: - -- -- Starting simple switch on 'DAHDI/2-1' -- Blacklisting number 201 - -- I do not believe that it is something own of the analogical telephone. Yesterday, exactly, I was testing with another telephone (of my work) to discard that it could be something of the house telephone, and it happens the same exactly. Making the changes in logger.conf to also see the dialing DTMF tones, they seem to be correctly passed: - -- -- Starting simple switch on 'DAHDI/2-1' [Jun 4 06:47:16] DTMF[8669]: channel.c:2229 __ast_read: DTMF end '*' received on DAHDI/2-1, duration 0 ms [Jun 4 06:47:16] DTMF[8669]: channel.c:2271 __ast_read: DTMF end accepted without begin '*' on DAHDI/2-1 [Jun 4 06:47:16] DTMF[8669]: channel.c:2282 __ast_read: DTMF end passthrough '*' on DAHDI/2-1 [Jun 4 06:47:16] DTMF[8669]: channel.c:2229 __ast_read: DTMF end '6' received on DAHDI/2-1, duration 0 ms [Jun 4 06:47:16] DTMF[8669]: channel.c:2271 __ast_read: DTMF end accepted without begin '6' on DAHDI/2-1 [Jun 4 06:47:16] DTMF[8669]: channel.c:2282 __ast_read: DTMF end passthrough '6' on DAHDI/2-1 [Jun 4 06:47:17] DTMF[8669]: channel.c:2229 __ast_read: DTMF end '0' received on DAHDI/2-1, duration 0 ms [Jun 4 06:47:17] DTMF[8669]: channel.c:2271 __ast_read: DTMF end accepted without begin '0' on DAHDI/2-1 [Jun 4 06:47:17] DTMF[8669]: channel.c:2282 __ast_read: DTMF end passthrough '0' on DAHDI/2-1 -- Blacklisting number cell_number [Jun 4 06:47:21] DEBUG[8669]: chan_dahdi.c:6244 ss_thread: waitfordigit returned 0... -- Hungup 'DAHDI/2-1' -- Starting simple switch on 'DAHDI/2-1' [Jun 4 06:47:26] DEBUG[8670]: chan_dahdi.c:6244 ss_thread: waitfordigit returned 0... -- Hungup 'DAHDI/2-1' - -- As I've commented in a previous message, after dial *60 (of *600 to Echo test), I obtain like a tone cut in three parts followed of a continuous tone, causing that I'm incapable to dial the extension completely. The waitfordigit appears after to hangup. The cell_number seems to be some number that I has dial previously. Testing again with a SIP extension, this problem does not happen. Also it draws attention to me that the DTMF has a duration of 0ms. It is peculiar... after to have a restart of Asterisk, I can dial without problems to *600. This is Asterisk log corresponding to the successful communication with the extension: - -- [Jun 4 23:03:30] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '*' received on DAHDI/2-1, duration 0 ms [Jun 4 23:03:30] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted without begin '*' on DAHDI/2-1 [Jun 4 23:03:30] DTMF[28905]: channel.c:2282 __ast_read: DTMF end passthrough '*' on DAHDI/2-1 [Jun 4 23:03:30] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '6' received on DAHDI/2-1, duration 0 ms [Jun 4 23:03:30] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted without begin '6' on DAHDI/2-1 [Jun 4 23:03:30] DTMF[28905]: channel.c:2282 __ast_read: DTMF end passthrough '6' on DAHDI/2-1 [Jun 4 23:03:31] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '0'
[asterisk-users] Transfer call from analog telephone
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I'm trying to doing a transfer from an analog extension to a SIP
extension but until the moment I was not successful.
I was testing both the recall key and uncomment the following
lines in the features.conf file:
blindxfer = #1
atxfer = *2
verifying previously that the extension uses the arguments tT with the
Dial application and to include the context featuremap in the context
in which I have defined the extensions (internal).
The telephone of the end with which the conversation is staying listens
the tones to try doing the transfer, but Asterisk does not give the dial
tone after *2 / #1 or the recall key.
I copy my configuration files after to have reverted the changes. If some
other data is necessary, don't doubt in consulting to me. The lines that I
added to the configuration files created in the installation are those that
are underneath DGB.
## /etc/asterisk/features.conf
[general]
parkext = 700 ; What extension to dial to park
parkpos = 701-720 ; What extensions to park calls on. These needs
to be
; numeric, as Asterisk starts from the start
position
; and increments with one for the next parked
call.
context = parkedcalls ; Which context parked calls are in
; (default is 45 seconds)
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
; one of: parked, caller, both (default is
caller)
; one of: callee, caller, both, no (default is
both)
; one of: callee, caller, both, no (default is
no)
; one of: callee, caller, both, no (default is
no)
; one of: callee, caller, both, no (default is
no)
; Defaults to 'first' available
; as long as the class is not set on the
channel directly
; using Set(CHANNEL(musicclass)=whatever) in
the dialplan
; (default is 3 seconds)
; feature activation (default is 1000 ms)
[featuremap]
[applicationmap]
## /etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/G2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually
1 or 0)
[default]
; DGB
[internal]
exten = _2xx,1,Dial(SIP/${EXTEN},15,tTm)
exten = _2xx,2,VoiceMail(${ext...@voicemail)
exten = _2xx,3,Playback(vm-goodbye)
exten = _2xx,4,Hangup
exten = *98,1,Answer
exten = *98,2,Wait(1)
exten = *98,3,VoiceMailMain(${caller...@voicemail)
exten = *98,4,Hangup
exten = *600,1,Answer
exten = *600,2,Playback(demo-echotest)
exten = *600,3,Echo
exten = *600,4,Playback(demo-echodone)
exten = *600,5,Hangup
exten = _9.,1,Dial(DAHDI/1/${EXTEN:1})
exten = _9.,2,Hangup
exten = 1010,1,Dial(DAHDI/2,15,tTm)
exten = 1010,2,Hangup
include = phones
[phones]
include = internal
[incoming]
exten = s,1,Dial(SIP/201,15,tTm)
exten = s,2,Hangup
## /etc/asterisk/chan_dahdi.conf
[trunkgroups]
[channels]
context=default
switchtype=national
signalling=fxo_ls
rxwink=300 ; Atlas seems to use long (250ms) winks
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
group=1
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
; DGB
language=es
defaultzone=es
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
inmediate=no
context=phones
signalling=fxo_ks
channel = 2 ; Telephone attached to port 2
context=incoming
signalling=fxs_ks ; Use FXS signalling for an FXS channel
channel = 1 ; PSTN attached to port 1
##
Which can be the problem or what configuration can be lacking?
Thanks in avance.
Regards,
Daniel
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[asterisk-users] Problem releasing call from a SIP extension
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! Making some changes in extensions.conf to test the incoming calls so that these are derived to a SIP extension, I found something that draws attention to me: if I test calling to my PSTN line from a mobile phone, when take the call from the SIP extension (softphone), if the mobile phone releases the call, sofphone do it too without problems, but if I release the call from sofphone, from the mobile phone I see that the call chronometer continues advancing as if the mobile phone not yet releases the call. Which can be the problem? Seeing in logs of the CLI, I observed the following thing: - --- alderamin*CLI -- Starting simple switch on 'DAHDI/1-1' [May 30 14:28:46] NOTICE[18535]: chan_dahdi.c:6830 ss_thread: Got event 18 (Ring Begin)... [May 30 14:28:47] NOTICE[18535]: chan_dahdi.c:6830 ss_thread: Got event 2 (Ring/Answered)... -- Executing [...@incoming:1] Dial(DAHDI/1-1, SIP/201|15|tT) in new stack -- Called 201 -- SIP/201-09243ea8 is ringing -- Nobody picked up in 15000 ms -- Executing [...@incoming:2] Hangup(DAHDI/1-1, ) in new stack == Spawn extension (incoming, s, 2) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' -- Starting simple switch on 'DAHDI/1-1' [May 30 14:29:11] NOTICE[18544]: chan_dahdi.c:6830 ss_thread: Got event 18 (Ring Begin)... [May 30 14:29:11] ERROR[18544]: callerid.c:564 callerid_feed: No start bit found in fsk data. [May 30 14:29:11] WARNING[18544]: chan_dahdi.c:6870 ss_thread: CallerID feed failed: Success [May 30 14:29:11] WARNING[18544]: chan_dahdi.c:6970 ss_thread: CallerID returned with error on channel 'DAHDI/1-1' -- Executing [...@incoming:1] Dial(DAHDI/1-1, SIP/201|15|tT) in new stack -- Called 201 -- SIP/201-09243ea8 is ringing -- Nobody picked up in 15000 ms -- Executing [...@incoming:2] Hangup(DAHDI/1-1, ) in new stack == Spawn extension (incoming, s, 2) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' -- Starting simple switch on 'DAHDI/1-1' [May 30 14:29:36] NOTICE[18554]: chan_dahdi.c:6830 ss_thread: Got event 18 (Ring Begin)... [May 30 14:29:36] ERROR[18554]: callerid.c:564 callerid_feed: No start bit found in fsk data. [May 30 14:29:36] WARNING[18554]: chan_dahdi.c:6870 ss_thread: CallerID feed failed: Success [May 30 14:29:36] WARNING[18554]: chan_dahdi.c:6970 ss_thread: CallerID returned with error on channel 'DAHDI/1-1' -- Executing [...@incoming:1] Dial(DAHDI/1-1, SIP/201|15|tT) in new stack -- Called 201 -- SIP/201-09243ea8 is ringing -- SIP/201-09243ea8 answered DAHDI/1-1 == Spawn extension (incoming, s, 1) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' - --- The lines that I'm using in the configuration file are the following: [incoming] exten = s,1,Dial(SIP/201,15,tT) exten = s,2,Hangup I think that as timeout of the call is 15 seconds and the mobile phone still continues calling, that causes that every 15 seconds it execute again a switch on 'DAHDI/1-1'. Can the message exited non-zero on 'DAHDI/1-1' have relation with the problem? I was testing calling from my cell phone to an analog telephone and if the other person hangs before I do it, I see that in the my cell phone the call even continues persisting so that if the person of the other endpoint take the earphone again after to hang, we can continue speaking :-D It will be some trick of the telephone companies to collect more with the unwary subscribers? :-D Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoh6HsACgkQZpa/GxTmHTca+wCfd7ogHaozBDc37DVnT0lrMmYU vYUAni1hp7irLmQNVYt0c3dz0vcaoYVa =UYO9 -END PGP SIGNATURE- ___ -- 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] Channels configuration with DAHDI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 El domingo 24 de mayo del 2009 a las 19:38:30 -0300, Daniel Bareiro escribió: Now it would remain to find the cause of why I cannot call from a SIP extension to an analog telephone. Perhaps it is by something related to the contexts in the mentioned configuration files? I forgot to copy the output that I obtain in the CLI when I call to a SIP extension: [May 25 19:22:57] NOTICE[4813]: chan_sip.c:14721 handle_request_invite: Call from '201' to extension '1010' rejected because extension not found. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkobGyoACgkQZpa/GxTmHTcYdgCfW8RUyUY5e4pbxs5xC/9Fcpp7 58UAnRAfj2eUL8ZAtvgUxIwHvCv2OXDM =o+On -END PGP SIGNATURE- ___ -- 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] Channels configuration with DAHDI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Tzafrir, Danny. El jueves 21 de mayo del 2009 a las 06:55:14 -0300, Tzafrir Cohen escribió: Mmmm... but I believe that it had done already in that order. In fact, I reviewed the existence of the module and it was in the directory. For that reasonI said that perhaps it was bug by the following thing: [May 20 20:49:07] WARNING[23599]: loader.c:359 load_dynamic_module: Error loading module 'chan_dadhi.so': /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: ^ No such file or directory [May 20 20:49:07] WARNING[23599]: loader.c:653 load_resource: Module 'chan_dadhi.so' could not be loaded. Apparently Asterisk is looking for the module using an incorrect name. Whatever happens, I compile Asterisk again but I got the same error message. Now that I see the error with a little more of thoroughness, it seems that when doing a copy/paste in the CLI, the name of the module was incorrect and for that reason I got that error message :-). Now get a different error: alderamin*CLI module unload chan_dahdi.so alderamin*CLI module load chan_dahdi.so [May 21 06:15:34] WARNING[25314]: chan_dahdi.c:1233 dahdi_open: Unable to specify channel 2: No such device or address [May 21 06:15:34] ERROR[25314]: chan_dahdi.c:7662 mkintf: Unable to open channel 2: No such device or address here = 0, tmp-channel = 2, channel = 2 [May 21 06:15:34] ERROR[25314]: chan_dahdi.c:11270 build_channels: Unable to register channel '2' In your configuration channel 2 is the first one, so this could be just about anything related to accessing dahdi / zaptel . Do you use dahdi or zaptel? What is the output of: cat /proc/zaptel/* cat /proc/dahdi/* # cat /proc/dahdi/* Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 WCTDM/4/0 RED 2 WCTDM/4/1 3 WCTDM/4/2 4 WCTDM/4/3 After to run dahdi_cfg -vvv: # cat /proc/dahdi/* Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 WCTDM/4/0 FXSKS RED 2 WCTDM/4/1 FXOKS 3 WCTDM/4/2 4 WCTDM/4/3 Two doubts related to this: * what means the word 'RED' associated to zero channel? I observed when I connect the telephone line here, the word 'RED' disappears. * I have the impression that the execution of this command is necessary for the correct detection of the modules. This execution is made automatically during bootstrapping of the operating system? Then I test the following thing in the CLI: alderamin*CLI dahdi show channel No such command 'dahdi show channel' (type 'help dahdi show' for other possible commands) alderamin*CLI module load chan_dahdi.so == Parsing '/etc/asterisk/chan_dahdi.conf': Found -- Registered channel 2, FXO Kewlstart signalling [May 24 15:04:54] WARNING[5306]: chan_dahdi.c:4090 handle_alarms: Detected alarm on channel 1: Red Alarm -- Registered channel 1, FXS Kewlstart signalling -- Automatically generated pseudo channel == Parsing '/etc/asterisk/users.conf': Found == Registered channel type 'DAHDI' (DAHDI Telephony Driver) == Manager registered action DAHDITransfer == Manager registered action ZapTransfer == Manager registered action DAHDIHangup == Manager registered action ZapHangup == Manager registered action DAHDIDialOffHook == Manager registered action ZapDialOffHook == Manager registered action DAHDIDNDon == Manager registered action ZapDNDon == Manager registered action DAHDIDNDoff == Manager registered action ZapDNDoff == Manager registered action DAHDIShowChannels == Manager registered action ZapShowChannels == Manager registered action DAHDIRestart == Manager registered action ZapRestart Loaded chan_dahdi.so = (DAHDI Telephony) alderamin*CLI alderamin*CLI alderamin*CLI alderamin*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault 1incomming es default 2phones es default Good, now it seems that it would be working... but is there any form to doing that this module is also automatically load when bootstrappiong of the operating system? I was looking for in the configuration files of example that are in /etc/asterisk but I did not find some reference. After this, I was doing some tests of connection calling from an SIP extension to the PSTN and this worked perfectly, but didn't get yet to connect an extension SIP with a conventional analog telephone to each other. If you can give me some guide line (unnecessary configurations or something that are lacking and could be useful) to know what to correct in this aspect, it would be very useful. From an analog telephone I can to call to a SIP extension, but from an SIP extension I cannot to call to the analog telephone. These are the configuration files
Re: [asterisk-users] Channels configuration with DAHDI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Tzafrir. El domingo 24 de mayo del 2009 a las 17:33:36 -0300, Tzafrir Cohen escribió: # cat /proc/dahdi/* Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 WCTDM/4/0 RED 2 WCTDM/4/1 3 WCTDM/4/2 4 WCTDM/4/3 After to run dahdi_cfg -vvv: # cat /proc/dahdi/* Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 WCTDM/4/0 FXSKS RED 2 WCTDM/4/1 FXOKS 3 WCTDM/4/2 4 WCTDM/4/3 Two doubts related to this: * what means the word 'RED' associated to zero channel? I observed when I connect the telephone line here, the word 'RED' disappears. As you guessed: it means that the line is connected to a working FXS (as I wrote in my previous mail). Perfect. * I have the impression that the execution of this command is necessary for the correct detection of the modules. It's not really detection. It is the DAHDI indicating you that the channel is actually configured. I spoke of 'detection' because after the execution of dahdi_cfg only is that 'cat /proc/dahdi/*' shows FXSKS and FXOKS. Perhaps, it would have been more correct to say the execution of this command is necessary to configure the channels signaling. This execution is made automatically during bootstrapping of the operating system? Yes. At boot you run /etc/init.d/dahdi which runs dahdi_cfg after the modules are loaded. There's a deprecated method of running dahdi_cfg as a post-load command of a module, but it is an ugly workaround that causes too many problems. Perfect. Then I test the following thing in the CLI: alderamin*CLI dahdi show channel No such command 'dahdi show channel' (type 'help dahdi show' for other possible commands) alderamin*CLI module load chan_dahdi.so == Parsing '/etc/asterisk/chan_dahdi.conf': Found -- Registered channel 2, FXO Kewlstart signalling [May 24 15:04:54] WARNING[5306]: chan_dahdi.c:4090 handle_alarms: Detected alarm on channel 1: Red Alarm -- Registered channel 1, FXS Kewlstart signalling -- Automatically generated pseudo channel == Parsing '/etc/asterisk/users.conf': Found == Registered channel type 'DAHDI' (DAHDI Telephony Driver) == Manager registered action DAHDITransfer == Manager registered action ZapTransfer == Manager registered action DAHDIHangup == Manager registered action ZapHangup == Manager registered action DAHDIDialOffHook == Manager registered action ZapDialOffHook == Manager registered action DAHDIDNDon == Manager registered action ZapDNDon == Manager registered action DAHDIDNDoff == Manager registered action ZapDNDoff == Manager registered action DAHDIShowChannels == Manager registered action ZapShowChannels == Manager registered action DAHDIRestart == Manager registered action ZapRestart Loaded chan_dahdi.so = (DAHDI Telephony) alderamin*CLI alderamin*CLI alderamin*CLI alderamin*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault 1incomming es default 2phones es default Good, now it seems that it would be working... but is there any form to doing that this module is also automatically load when bootstrappiong of the operating system? I was looking for in the configuration files of example that are in /etc/asterisk but I did not find some reference. Yes. If /etc/inint.d/asterisk is run after /etc/init.d/dahdi (which should happen with default installs of Asterisk and DAHDI) this should be the case. This is consistent: /etc/rc2.d/S50asterisk /etc/rc2.d/S15dahdi I tried only leaving uncommented in the file /etc/dahdi/modules the line 'wctdm' and after to reboot the operating system, both the card as FXS/FXO channels signaling was detected without problems. Now it would remain to find the cause of why I cannot call from a SIP extension to an analog telephone. Perhaps it is by something related to the contexts in the mentioned configuration files? Thank for your reply. Regads, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoZzKMACgkQZpa/GxTmHTfPcgCfUL5muSBicoU3bAdDKC0ZSkzM 7z4AnRg2YWyeEM5CEhYeuoj3RAQyH4K3 =Ywas -END PGP SIGNATURE- ___ -- 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] Channels configuration with DAHDI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 El miércoles 20 de mayo del 2009 a las 21:19:18 -0300, Daniel Bareiro escribió: I load the modules wctdm and dahdi. But when I execute in Asterisk CLI dahdi show channels, I get the following error message: No such command 'dahdi show channels' (type 'help dahdi show' for other possible commands) Try running: asterisk -r and in that prompt: module unload chan_dadhi.so module load chan_dadhi.so and tell us the output you got. # asterisk -r Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Connected to Asterisk 1.4.24.1 currently running on alderamin (pid = 19777) Verbosity is at least 7 alderamin*CLI alderamin*CLI module unload chan_dadhi.so alderamin*CLI module load chan_dadhi.so [May 20 17:52:19] WARNING[10345]: loader.c:359 load_dynamic_module: Error loading module 'chan_dadhi.so': /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: No such file or directory [May 20 17:52:19] WARNING[10345]: loader.c:653 load_resource: Module 'chan_dadhi.so' could not be loaded. alderamin*CLI Mmmm... it would seem to be a bug: /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: No such file or directory Sounds like DAHDI was installed/compiled *after* Asterisk was compiled. Recompile Asterisk again and make sure /usr/lib/asterisk/modules/chan_dahdi.so is created when you make install. Mmmm... but I believe that it had done already in that order. In fact, I reviewed the existence of the module and it was in the directory. For that reasonI said that perhaps it was bug by the following thing: [May 20 20:49:07] WARNING[23599]: loader.c:359 load_dynamic_module: Error loading module 'chan_dadhi.so': /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: ^ No such file or directory [May 20 20:49:07] WARNING[23599]: loader.c:653 load_resource: Module 'chan_dadhi.so' could not be loaded. Apparently Asterisk is looking for the module using an incorrect name. Whatever happens, I compile Asterisk again but I got the same error message. Now that I see the error with a little more of thoroughness, it seems that when doing a copy/paste in the CLI, the name of the module was incorrect and for that reason I got that error message :-). Now get a different error: alderamin*CLI module unload chan_dahdi.so alderamin*CLI module load chan_dahdi.so [May 21 06:15:34] WARNING[25314]: chan_dahdi.c:1233 dahdi_open: Unable to specify channel 2: No such device or address [May 21 06:15:34] ERROR[25314]: chan_dahdi.c:7662 mkintf: Unable to open channel 2: No such device or address here = 0, tmp-channel = 2, channel = 2 [May 21 06:15:34] ERROR[25314]: chan_dahdi.c:11270 build_channels: Unable to register channel '2' Can this be due to something incorrect in the configurations that I mentioned in a previous post? Whatever happens, could that be the cause by which the command 'dahdi show status' is not found? Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoVIU8ACgkQZpa/GxTmHTcLaACZAdNfwO5l1ZRZlX4swvJJhpjI 27MAoJEK3SOw3nsTFgU0TgYJ9NhmbrUz =ln/i -END PGP SIGNATURE- ___ -- 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] Channels configuration with DAHDI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! Days ago I bought a OpenVox A400P card with a port FXS and another FXO that I am testing with my Asterisk installation in my house. I'm using Asterisk 1.4.24.1 with DAHDI Linux 2.1.0.4 and DAHDI Tools 2.1.0.2 on Debian GNU/Linux Lenny. I was reading The future of telephony and this [1] document looking for information about how to configure both types of channels. Hardware is recognized without problems by operating system and DAHDI: # lspci [...] 00:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface [...] # cat /proc/dahdi/* Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: HRtimer) 1 (MASTER) Span 2: WCTDM/4 Wildcard TDM400P REV E/F Board 5 1 WCTDM/4/0 FXSKS RED 2 WCTDM/4/1 FXOKS 3 WCTDM/4/2 4 WCTDM/4/3 # dahdi_hardware pci::00:0a.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F # dahdi_cfg -vvv DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) 2 channels to configure. In the configuration files I made the following modifications: /etc/dahdi/system.conf: # DGB - 20090518 fxoks=2 fxsks=1 DGB #[channels] #language=en #context=incoming #signalling=fxs_ks #usecallerid=yes #hidecallerid=no #callwaiting=yes #callwaitingcallerid=yes #threewaycalling=yes #transfer=yes #cancallforward=yes #callreturn=yes #echocancel=yes #echocancelwhenbridged=yes #rxgain=0.0 #txgain=0.0 #group=1 #pickupgroup=1 #immediate=yes #musiconhold=default channel = 1 DGB /etc/asterisk/chan_dahdi.conf: ; DGB - 20090518 group=0 signaling=fxo_ks channel = 2 signaling=fxs_ks channel = 1 I load the modules wctdm and dahdi. But when I execute in Asterisk CLI dahdi show channels, I get the following error message: No such command 'dahdi show channels' (type 'help dahdi show' for other possible commands) I suppose that beyond not to have added in /etc/asterisk/extensions.conf something like: exten = s,1,Wait(1) ; Wait a second, just for fun exten = s,n,Answer ; Answer the line exten = s,n,Dial(dahdi/2) ; zap has been changed to dahdi exten = s,n,Hangup exten = 1,1,Dial(dahdi/2|60|m(default)) ; zap has been changed to dahdi exten = 1,2,hangup the command would have to exist. Which can be the problem? Thanks in advance. Regards, Daniel [1] http://bbs2.chinaunix.net/archiver/tid-1289253.html -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoT1Z4ACgkQZpa/GxTmHTfqwACfdDW6qLmmQgSdlI2MyK0W/kjK B+EAn15G9AqYKNEWRb+p3OyZW0LSXIjf =LBis -END PGP SIGNATURE- ___ -- 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] Channels configuration with DAHDI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Tzafrir. El miércoles 20 de mayo del 2009 a las 10:00:46 -0300, Tzafrir Cohen escribió: On Wed, May 20, 2009 at 07:03:15AM -0300, Daniel Bareiro wrote: Hint: you don't need to set 'signalling' for analog channels. Or just set it explicitly to auto. This is for Asterisk = 1.6.0 . Simply reduces the complication a bit... Thanks for the tip. I will remember it for when I use Asterisk 1.6 :-) I load the modules wctdm and dahdi. But when I execute in Asterisk CLI dahdi show channels, I get the following error message: No such command 'dahdi show channels' (type 'help dahdi show' for other possible commands) Try running: asterisk -r and in that prompt: module unload chan_dadhi.so module load chan_dadhi.so and tell us the output you got. # asterisk -r Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Connected to Asterisk 1.4.24.1 currently running on alderamin (pid = 19777) Verbosity is at least 7 alderamin*CLI alderamin*CLI module unload chan_dadhi.so alderamin*CLI module load chan_dadhi.so [May 20 17:52:19] WARNING[10345]: loader.c:359 load_dynamic_module: Error loading module 'chan_dadhi.so': /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: No such file or directory [May 20 17:52:19] WARNING[10345]: loader.c:653 load_resource: Module 'chan_dadhi.so' could not be loaded. alderamin*CLI Mmmm... it would seem to be a bug: /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: No such file or directory Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoUb1QACgkQZpa/GxTmHTdOfgCeKIw4uNyP6MOboezpFBrYBB2B H4YAoIm+CfhEpWMBFyg2cbUyxNIh6ivL =JgUc -END PGP SIGNATURE- ___ -- 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] Channels configuration with DAHDI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Dave. El miércoles 20 de mayo del 2009 a las 18:12:04 -0300, Dave Fullerton escribió: I load the modules wctdm and dahdi. But when I execute in Asterisk CLI dahdi show channels, I get the following error message: No such command 'dahdi show channels' (type 'help dahdi show' for other possible commands) Try running: asterisk -r and in that prompt: module unload chan_dadhi.so module load chan_dadhi.so and tell us the output you got. # asterisk -r Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Connected to Asterisk 1.4.24.1 currently running on alderamin (pid = 19777) Verbosity is at least 7 alderamin*CLI alderamin*CLI module unload chan_dadhi.so alderamin*CLI module load chan_dadhi.so [May 20 17:52:19] WARNING[10345]: loader.c:359 load_dynamic_module: Error loading module 'chan_dadhi.so': /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: No such file or directory [May 20 17:52:19] WARNING[10345]: loader.c:653 load_resource: Module 'chan_dadhi.so' could not be loaded. alderamin*CLI Mmmm... it would seem to be a bug: /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: No such file or directory Sounds like DAHDI was installed/compiled *after* Asterisk was compiled. Recompile Asterisk again and make sure /usr/lib/asterisk/modules/chan_dahdi.so is created when you make install. Mmmm... but I believe that it had done already in that order. In fact, I reviewed the existence of the module and it was in the directory. For that reasonI said that perhaps it was bug by the following thing: [May 20 20:49:07] WARNING[23599]: loader.c:359 load_dynamic_module: Error loading module 'chan_dadhi.so': /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: ^ No such file or directory [May 20 20:49:07] WARNING[23599]: loader.c:653 load_resource: Module 'chan_dadhi.so' could not be loaded. Apparently Asterisk is looking for the module using an incorrect name. Whatever happens, I compile Asterisk again but I got the same error message. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoUnkMACgkQZpa/GxTmHTfKeACffr1Q2vgJnyDVrA+hCN3DtHd5 e4UAoJUSl6JjjMCX+SiUA2/cyHJhwtfN =wCSV -END PGP SIGNATURE- ___ -- 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] Beginning to use Asterisk and tests with extensions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello Daniel, Hi Dana. You will find the information at http://www.voip-info.org/ and http://oreilly.com/catalog/9780596510480/ (.PDF downloadable from the Online Book link) very useful. I have the second edition that covers Asterisk 1.4 and it seems interesting. You made me remember that I had downloaded it the last year, although just now I have more time to dedicate to Asterisk. The fact of to have already installed it is an important step :-) The asterisk package by itself should be adequate for SIP/IAX calls. I don't think you need libpri unless you are planning on connecting asterisk to a digital connection such as ISDN or a PRI. You will need Zaptel (for Asterisk versions 1.2,1.4) or DAHDI (Asterisk versions =1.6) if you choose to install an internal card (OpenVOX, Digium, Sangoma, etc.) I do not know if or how well this will work with a VM. Thanks for the indication. According to I saw in the site of Asterisk[1], only make reference to DAHDI for Asterisk 1.4, but according to which you say to me, both can be used. My idea is to buy an ATA to connect a conventional telephone and make tests of communication between it and softphone. The idea by which I thought about using an ATA is because I am not sure with my version of KVM (KVM-62) can make PCI pass through. But with the ATA must not have problem. Having this in mind, I installed the packages dahdi-linux-2.1.0.4.tar.gz and dahdi-tools-2.1.0.2.tar.gz having loaded only the module dahdi_dummy and so far commenting all that appear in /etc/dahdi/modules. I suggest testing your SIP softphone with the Echo() and/or Playback() dialplan applications before attempting to call another softphone/hardphone/etc. This will allow you to confirm that the one endpoint functions properly before adding more complexity by calling another endpoint. I was testing and sometimes with Echo() and MusicOnHold the sound is broken. Is there some form to solve this? some things that allow you to call a conventional telephone: an ATA with an FXS port an internal card (such as OpenVOX, Digium, Sangoma) with an FXS port call a conventional phone number through the PSTN (below) To connect to the PSTN you can use any of: an ATA with an FXO port (plug an analog phone line into it) internal card with an FXO port (also to plug an analog phone line in) account with an ITSP (there is occasionally discussion on the list about advantages/issues/opinions/and flames with various ITSPs - google site:lists.digium.com ITSP) [...] I believe that with the example I understood a little better how it works. As it mentioned above, I am thinking about buying a Linksys SPA3102 to make both internals and with PSTN tests. Hope that gets you going in the right direction. http://www.voipsupply.com/ is a good source to see what equipment is generally available to end users. Thanks for your reply and by all the references and examples that you provided to me. Regards, Daniel [1] http://www.asterisk.org/downloads -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoHNXIACgkQZpa/GxTmHTccUACfbHY+st10rhsqrsZnE9SJLZrV hFQAnR5Y85XQQr7Jm1wWzD106qxNkd4g =EYud -END PGP SIGNATURE- ___ -- 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] Beginning to use Asterisk and tests with extensions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 El domingo 10 de mayo del 2009 a las 17:12:51 -0300, Daniel Bareiro escribió: I suggest testing your SIP softphone with the Echo() and/or Playback() dialplan applications before attempting to call another softphone/hardphone/etc. This will allow you to confirm that the one endpoint functions properly before adding more complexity by calling another endpoint. I was testing and sometimes with Echo() and MusicOnHold the sound is broken. Is there some form to solve this? Investigating a little more in Internet, it seems that the expression used in english for this is choppy sound. According to it seems, the problem was of the side of the client and I could solve it of the form commented here [1] (softphone I use is Twinkle). The problem with this is that although Twinkle now work, I can't have anything else running that uses sound because then the audio is blocked. Regards, Daniel [1] http://www.lynchconsulting.com.au/blog/index.cfm/2008/11/6/Choppy-sound-on-Twinkle-Softphone-on-Ubuntu-Linux -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoHX1UACgkQZpa/GxTmHTezugCdFqm+vBOIwsTGUYS7ecTT7DcP TX4An0kU/YtK5w1CDMEln8vabOzRJNC4 =iaWE -END PGP SIGNATURE- ___ -- 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] Beginning to use Asterisk and tests with extensions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! This is my first message to the list/newsgroup. This weekend and after to have fought by some time with my soundcard with respecto to the voice capture, after assuring to have solved that problem, I installed Asterisk on Debian GNU/Linux Lenny. I made my installation on a KVM virtual machine. In order to begin and according to I could see on the basis of which I was reading in Internet, to make a basic installation initially it would be enough with the packages 'asterisk' and 'libpri', reason why those were these that I installed at the moment. But correct to me, if I'm mistaken, please. However, the following basic step would be to test with extensions and since in my house I only have a PC that use like workstation, is some complicated to test of calls :-) Whatever happens, I installed Twinkle from Debian GNU/Linux repositories. But to make a valid test would need another PC with softphone or something that allows me to call to a conventional telephone. For this I, read in some documents that the ATAs are mentioned (bah, I believe that the denomination ATA is something own of CISCO and perhaps most appropriate is to call it as Adapters for Analogical Telephones), that allows to connect a conventional telephone to a VoIP network of way to be able to send and to receive calls having an Ethernet connector to connect it to the LAN. What not yet it is clear to me of these ATAs is how they works. I have understood that it have its own IP and the one of PBX server, but if we have, for example, two FXS ports connecting to each of them to a conventional telephone, in the documentation that I found at the moment is not mentioned some way to associate the ports of the conventional telephones with a number of extension so that the ATA knows how to route an incoming call. The other alternative is to use a OpenVOX card, for example, but I'm not sure if this solution is worth to me because if I install it in the PC where I have the virtual machine with the Asterisk, I'm not sure if the KVM virtual machine can access to that underlying hardware. Thanks in advance and with the time I hope to be gaining knowledge also to be able to make some contribution to the list/newsgroup. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoAD3QACgkQZpa/GxTmHTeL/QCZAa0cy77+5YIHAx5wRvDeSvTd 01EAn2yPyHuyeaabkqLL9yW4S0rb2USx =2TBa -END PGP SIGNATURE- ___ -- 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
