Re: [asterisk-users] Getting calee audio in Asterisk (real time)
Just FYI how I solved this: I figured out that JACK_HOOK`ing for open channel does not connect input and output ports. So instead of *CLI core set chanvar SIP/poly1-ab23jadf234 JACK_HOOK(manipulate) on you shoud use: *CLI core set chanvar SIP/poly1-ab23jadf234 JACK_HOOK(manipulate,i(SIP/poly1-ab23jadf234:input),o(SIP/poly1-ab23jadf234:output)) on Then all works fine and you get leg B's channel. -- Forwarded message -- From: Motiejus Jakštys desired@gmail.com Date: 2010/5/5 Subject: Re: Getting calee audio in Asterisk (real time) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Update: I thought this may be the solution: *CLI core set chanvar SIP/poly1-ab23jadf234 JACK_HOOK(manipulate) on (For 1.6.2 it's dialplan set chanvar SIP/poly1-ab23jadf234 JACK_HOOK(manipulate) on ) Source: voip-info.org The command opens two jack ports: Channel:input and channel:output. At once command is executed, sound on the caller is gone. Question: what should this CLI command do in reality? Is it a bug or expected behaviour? Then I connect those two ports hoping it will return the sound to the caller: jack_connect SIP/PBX2-000d:output SIP/PBX2-000d:input Then the calee hears garbled sound. Sample of all process is here. It is recorded by MixMonitor on the machine where jack takes process. Asterisk 1.6.2.6 (upgrading/downgrading/patching is not a problem). Waiting for your suggestions... Maybe I can do this in totally different approach? Regards Motiejus Jakštys http://m.jakstys.lt/ 2010/5/5 Motiejus Jakštys desired@gmail.com Hello, I need to capture calee's audio in real-time in order to capture operator messages (I've written sound recognition software that works with Jack: http://github.com/Motiejus/SoundPatty/). Jack does the following: Incoming call audio - audio in to jack, audio out from jack - current Asterisk application Outgoing call audio - current Asterisk application However, I need vica-versa: Incoming call audio - current Asterisk application Outgoing call audio - Audio from jack, Audio into Jack - current Asterisk application or at least Incoming call audio - current Asterisk application Audio to jack - current Asterisk application Outgoing call audio - current Asterisk application Any idea how I could accomplish this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Issues with remote call setup
Hello list, I would like to seek your expert opinion on a setup I am trying as part of my research. I have not been able to successfully make a call so far. In my setup, I use two laptops that are interconnected by means of a stand-alone IS1581 switch. Thus there is no LAN involved. I have assigned static IPs to the two laptops, say 10.0.0.1 and 10.0.0.2. I have installed Asterisk 1.6.2.6 and Zoiper classic (free version)on 10.0.0.1. On 10.0.0.2, I have installed Zoiper classic (free version). On 10.0.0.1, user1 has been configured on Zoiper. The domian for user1 has been set to 127.0.0.1. STUN has been disabled and SIP port is set to 5061. On 10.0.0.2, user3 has been configured on Zoiper. The domain for user3 has been set to 10.0.0.1. In addition, use of outbound proxy has been enabled for user3, and the address of the outbound proxy has been set to 10.0.0.1. STUN has been disabled and SIP port is set to 5061. As part of the asterisk configuration in 10.0.0.1, the following entries have been made in sip.conf: [general] context=default udpbindaddr=0.0.0.0 bindport=5060 srvlookup=no language=en contactpermit=127.0.0.1/255.255.255.0 contactpermit=10.0.0.2/255.255.255.0 sipdebug=yes allowsubscribe=no localnet=10.0.0.1/255.255.255.0 localnet=10.0.0.2/255.255.255.0 nat=never allowexternaldomains=no domain=10.0.0.1 matchexterniplocally=yes autodomain=yes directmedia=yes disallow=all allow=gsm allow=ulaw allow=alaw ;entry for phones [100] type=friend context=phones host=dynamic [102] type=friend context=phones host=dynamic ;entry for users [user1] type=friend context=on_this_system secret=password regcontext=on_this_system regexten=100 usereqphone=no host=dynamic nat=no [user3] type=friend context=on_that_system secret=password regcontext=on_that_system regexten=102 usereqphone=no host=dynamic nat=no And the following entries have been made in extensions.conf: [general] static=no writeprotect=no autofallthrough=no [default] [phones] include=internal include=remote [internal] exten = 100,1,Dial(SIP/user1, 25) exten = 100,n,Playback(vm-isunavail.gsm) exten = 100,n,Hangup() [remote] exten = 100,1,Dial(SIP/user3, 25) exten = 100,n,Playback(vm-isunavail.gsm) exten = 100,n,Hangup() [on_this_system] include=internal [on_that_system] include=remote With the above configuration, I am able to successfully register both the users with the asterisk server running on 10.0.0.1. However, when either user tries to call the other user's supposed extension, the call fails with the message no route to destination on Zoiper. But a loopback works successfully on both the laptops. Thus user1 can call itself, so can user3. Upon examining the SIP message logs on Wireshark, I could see that when a user on one laptop (Zoiper) tries to call the user on the other or attempts a loopback (call itself) by means of an extesion, the INVITE message is sent to the AOR sip:extension@domain. Thus when user1 tries to call user3 by dialing extension 102 on Zoiper, an INVITE is constructed with the AOR sip:1...@127.0.0.1. It would be useful to point out that 127.0.0.1 corresponds to 10.0.0.1, on which the asterisk server is running and with which user3 has been registered. This results in the asterisk server returning 404 Not Found response. In this case, I see the asterisk server is not able to map the dialed number to a registered user. In contrast, when a user tries to call itself, the call is connected, becuase asterisk server is able to map the dialed number to the registered user, as evinced in the logs. For example, user3 can dial 102 and the call gets connected. I have spent quite sometime debugging this without success. My inexperience is also a factor, as I am relatively new to Asterisk. Based on the above information, I would really appreciate if the experts in the list could point to the root cause of the problem. Thanks in advance. best regards Vin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Getting presence working in 1.6.2
I am running asterisk 1.6.2.6 and have configured hints for our extensions and have a couple of Aastra 6755i test phones. The phones register fine but 'core show hints' shows the lines as idle even if they are in use. I read the wiki and see mention about needing to set call-limit in asterisk 1.4 but that has been depreciated in 1.6 so what is the way it should be done in 1.6? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Dropping
On Sun, 2010-05-02 at 09:52 -0400, Dan Journo wrote: Hi Bob, Thanks for that. Is there any way I can make the task run in the background and free up the console? Also so that I can disconnect my ssh session without losing the task. Thanks Dan Matthieu NICAISE mentioned screen which should work. Another way would be to activate the script through cron: 1. create a script that does a few pings and e-mails the results. 2. activate the script with cron as often as needed. Once this is setup, you can quit your ssh access to the remote server. Contact me offlist if you need more information. Best regards, Bob Smither -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Getting presence working in 1.6.2
I read the wiki and see mention about needing to set call-limit in asterisk 1.4 but that has been depreciated in 1.6 so what is the way it should be done in 1.6? I set callcounter=yes in sip.conf. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: NAT in SPA922
On Thu, May 6, 2010 at 8:14 PM, Vineet Bhojnagarwala vbho...@gmail.com wrote: Alternatively, if using normal vlans, this can also be achieved by enabling access list on the switch and restrict traffic flows. Generally this is done on a layer 3 switch, don't think it will support on your switch model. That is correct. In order to do this on a 2950, you will need a router behind this to be the gateway for each vlan. (On Cisco equipment you'd need to create a subinterface for each vlan (i.e. FastEthernet 0.xxx) where xxx is your vlan number. Then you can set each port up to be a trunk port on the 2950, but specify the native vlan on the port as the PC vlan # and allow the Vlan # for the phone vlan. So something like: switchport mode trunk switchport trunk native vlan [pc vlan #] switchport trunk allowed vlan [pc vlan #],[phone vlan #] Then you will have to create access-lists on the router to block intra-VLAN traffic. This can also be all done on a Layer 3 switch (like the Cisco 3550), by defining each VLAN as an interface: interface VLAN 100 description Phone VLAN ip address 192.168.100.1 255.255.255.0 ! interface VLAN 101 description Customer 1 VLAN ip address 192.168.101.1 255.255.255.0 ! etc.. then your ports will look like: interface FastEthernet 0/2 description customer 1 port switchport mode trunk switchport trunk encapsulation dot1q switchport trunk native vlan 101 switchport trunk allowed vlan 100,101 ! Then you'll need access lists to prevent the intra-vlan traffic.. -- James Rgds, Vineet Bhojnagarwala RCDD, NTS, OSP Spear Networks Pvt Ltd Integration Consultancy +91-9831436607 On May 7, 2010, at 8:39 AM, Vineet Bhojnagarwala vbho...@gmail.com wrote: I think this is a motel kind of situation and a PVLAN serves the situation right. Put all the ipphones in the voice vlan as suggested, make a seperate isolated vlan for the PCs, this will restrict traffic between the clients. Rgds, Vineet Bhojnagarwala RCDD, NTS, OSP Spear Networks Pvt Ltd Integration Consultancy +91-9831436607 On May 6, 2010, at 11:30 PM, David White david.wh...@watchguard.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Noah Miller Sent: Thu 5/6/2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: NAT in SPA922 It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Put each PC in its own VLAN. Keep all the phones in one VLAN. Although having a $30 router in each room hanging off the phone would accomplish what you want also. Take j's suggestion to use VLANs. This is not a good situation for NAT. Cisco 2950's can do VLANs. to be clear, the only way this will work with the PCs is if each PC vlan is *also* a unique ip subnet (else how do all the vlans access a common default gw?) place the phones in a voice vlan, and the phone problem is solved. as for the PC isolation, you might get better feedback on a cisco or other networking forum. -david -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and gnokii on same server: scratchy sound
Hi, Has anyone tried to use gnokii to send/receive SMS messages via serial or USB with AT commands while running Asterisk? Some of my calls have a scratchy sound once in a while. It doesn't seem to be due to packet loss but some kind of interference (CPU is ok, etc.). I've noticed some coincidence in time between this scratchy sound and the gnokii process. I have a bash script that calls gnokii periodically to send/receive messages. The bad audio quality does not *always* appear when the gnokii process is up but just *sometimes*. If I stop my script, thus gnokii, it seems that audio quality is fine overall. What I still don't quite understand is who's responsible for this audio problem: gnokii itself (I don't think so), the GSM radio signal nearby (about 2 meters) or the data sent through the serial port/cable. The third explanation is the most probable but I'd like to know other people's opinions. Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] OT: NAT in SPA922
On May 7, 2010, at 8:03, James Lamanna jlama...@gmail.com wrote: On Thu, May 6, 2010 at 8:14 PM, Vineet Bhojnagarwala vbho...@gmail.com wrote: Alternatively, if using normal vlans, this can also be achieved by enabling access list on the switch and restrict traffic flows. Generally this is done on a layer 3 switch, don't think it will support on your switch model. That is correct. In order to do this on a 2950, you will need a router behind this to be the gateway for each vlan. (On Cisco equipment you'd need to create a subinterface for each vlan (i.e. FastEthernet 0.xxx) where xxx is your vlan number. Then you can set each port up to be a trunk port on the 2950, but specify the native vlan on the port as the PC vlan # and allow the Vlan # for the phone vlan. So something like: switchport mode trunk switchport trunk native vlan [pc vlan #] switchport trunk allowed vlan [pc vlan #],[phone vlan #] Then you will have to create access-lists on the router to block intra-VLAN traffic. This can also be all done on a Layer 3 switch (like the Cisco 3550), by defining each VLAN as an interface: interface VLAN 100 description Phone VLAN ip address 192.168.100.1 255.255.255.0 ! interface VLAN 101 description Customer 1 VLAN ip address 192.168.101.1 255.255.255.0 ! etc.. then your ports will look like: interface FastEthernet 0/2 description customer 1 port switchport mode trunk switchport trunk encapsulation dot1q switchport trunk native vlan 101 switchport trunk allowed vlan 100,101 ! Then you'll need access lists to prevent the intra-vlan traffic.. I lied. You don't need access-lists in this case with the allowed vlan statement. -- James Rgds, Vineet Bhojnagarwala RCDD, NTS, OSP Spear Networks Pvt Ltd Integration Consultancy +91-9831436607 On May 7, 2010, at 8:39 AM, Vineet Bhojnagarwala vbho...@gmail.com wrote: I think this is a motel kind of situation and a PVLAN serves the situation right. Put all the ipphones in the voice vlan as suggested, make a seperate isolated vlan for the PCs, this will restrict traffic between the clients. Rgds, Vineet Bhojnagarwala RCDD, NTS, OSP Spear Networks Pvt Ltd Integration Consultancy +91-9831436607 On May 6, 2010, at 11:30 PM, David White david.wh...@watchguard.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Noah Miller Sent: Thu 5/6/2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: NAT in SPA922 It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Put each PC in its own VLAN. Keep all the phones in one VLAN. Although having a $30 router in each room hanging off the phone would accomplish what you want also. Take j's suggestion to use VLANs. This is not a good situation for NAT. Cisco 2950's can do VLANs. to be clear, the only way this will work with the PCs is if each PC vlan is *also* a unique ip subnet (else how do all the vlans access a common default gw?) place the phones in a voice vlan, and the phone problem is solved. as for the PC isolation, you might get better feedback on a cisco or other networking forum. -david -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video in Skype for Asterisk
On 05/07/2010 12:23 AM, Richard Kenner wrote: Is there anything special that has to be done to make video calls work? Yeah... Skype needs to add video support to the Skype engine that SFA uses. It doesn't seem to work for me (no video). That's right. It's not supported. What CODECS are supported? No video codecs are supported; Skype clients only support VP7 and H.264 (most of them VP7), so it's not clear what is going to be possible once SFA does have video support. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hash Dial Pattern Problems
I changed the dial pattern to %23|XXX and dialed #1234567. I was able to trigger activity in the CLI: Connected to Asterisk 1.2.1 currently running on aikphone (pid = 29352) Verbosity is at least 22 -- Executing Macro(SIP/3000-ca1c, dialout-trunk|3|3643873|) in new stack -- Executing GotoIf(SIP/3000-ca1c, 1?3:2)) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/3000-ca1c, user-callerid) in new stack -- Executing DBget(SIP/3000-ca1c, AMPUSER=DEVICE/3000/user) in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=3000/user -- DBget: set variable AMPUSER to 3000 -- Executing DBget(SIP/3000-ca1c, AMPUSERCIDNAME=AMPUSER/3000/cidname) in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=3000/cidname -- DBget: set variable AMPUSERCIDNAME to Augusta I.T Tes -- Executing GotoIf(SIP/3000-ca1c, 0?5) in new stack -- Executing SetCallerID(SIP/3000-ca1c, Augusta I.T Tes 3000) in new stack -- Executing NoOp(SIP/3000-ca1c, Using CallerID Augusta I.T Tes 3000) in new stack -- Executing Macro(SIP/3000-ca1c, record-enable|3000|OUT) in new stack -- Executing GotoIf(SIP/3000-ca1c, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/3000-ca1c, recordingcheck|20100507-082747|1273235267.398) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20100507-082747|1273235267.398: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/3000-ca1c, No recording needed) in new stack -- Executing Macro(SIP/3000-ca1c, outbound-callerid|3) in new stack -- Executing DBget(SIP/3000-ca1c, USEROUTCID=AMPUSER/3000/outboundcid) in new stack -- DBget: varname=USEROUTCID, family=AMPUSER, key=3000/outboundcid -- DBget: set variable USEROUTCID to -- Executing GotoIf(SIP/3000-ca1c, 1?4) in new stack -- Goto (macro-outbound-callerid,s,4) -- Executing GotoIf(SIP/3000-ca1c, 1?6) in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing NoOp(SIP/3000-ca1c, CallerID set to Augusta I.T Tes 3000) in new stack -- Executing SetGroup(SIP/3000-ca1c, OUT_3) in new stack -- Executing CheckGroup(SIP/3000-ca1c, ) in new stack -- Executing SetVar(SIP/3000-ca1c, DIAL_NUMBER=3643873) in new stack -- Executing SetVar(SIP/3000-ca1c, DIAL_TRUNK=3) in new stack -- Executing AGI(SIP/3000-ca1c, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar(SIP/3000-ca1c, OUTNUM=3643873) in new stack -- Executing Cut(SIP/3000-ca1c, custom=OUT_3|:|1) in new stack -- Executing GotoIf(SIP/3000-ca1c, 0?16) in new stack -- Executing Dial(SIP/3000-ca1c, IAX2/augusta/3643873) in new stack -- Called augusta/3643873 -- Call accepted by 192.168.1.10 (format ulaw) -- Format for call is ulaw -- IAX2/augusta-16384 is making progress passing it to SIP/3000-ca1c -- Hungup 'IAX2/augusta-16384' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/3000-ca1c' in macro 'dialout-trunk' == Spawn extension (from-internal, %233643873, 1) exited non-zero on 'SIP/3000-ca1c' -- Executing Macro(SIP/3000-ca1c, hangupcall) in new stack -- Executing ResetCDR(SIP/3000-ca1c, w) in new stack -- Executing NoCDR(SIP/3000-ca1c, ) in new stack -- Executing Wait(SIP/3000-ca1c, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/3000-ca1c' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3000-ca1c' It stripped the hash and passed the number through the IAX2 trunk. I am just getting a all circuits are busy. Thanks, David On Wed, May 5, 2010 at 6:53 PM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! I set: sip debug peer 3000 (my test extension) and dialed #3643873 Your X-Lite softphone actually calls %233643873 and not #3643873. You would need to check the SIP RFCs in order to find out if Asterisk is behaving correctly here by not decoding %23 as #. In the meanwhile you could try to add the extension %233643873 to your dialplan, or find out if you can configure the way X-Lite handles the # within the dialstring. To: #3643873sip:%233643...@192.168.2.10sip%3a%25233643...@192.168.2.10 ... User-Agent: X-Lite release 1104o stamp 56125 (telephone-event) Looking for %233643873 in from-internal (domain ... SIP/2.0 404 Not Found Via: SIP/2.0/UDP Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list
Re: [asterisk-users] Hash Dial Pattern Problems
Now what does the 1.4 side (CLI) look like when you do this call? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel Sent: Friday, May 07, 2010 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hash Dial Pattern Problems I changed the dial pattern to %23|XXX and dialed #1234567. I was able to trigger activity in the CLI: Connected to Asterisk 1.2.1 currently running on aikphone (pid = 29352) Verbosity is at least 22 -- Executing Macro(SIP/3000-ca1c, dialout-trunk|3|3643873|) in new stack -- Executing GotoIf(SIP/3000-ca1c, 1?3:2)) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/3000-ca1c, user-callerid) in new stack -- Executing DBget(SIP/3000-ca1c, AMPUSER=DEVICE/3000/user) in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=3000/user -- DBget: set variable AMPUSER to 3000 -- Executing DBget(SIP/3000-ca1c, AMPUSERCIDNAME=AMPUSER/3000/cidname) in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=3000/cidname -- DBget: set variable AMPUSERCIDNAME to Augusta I.T Tes -- Executing GotoIf(SIP/3000-ca1c, 0?5) in new stack -- Executing SetCallerID(SIP/3000-ca1c, Augusta I.T Tes 3000) in new stack -- Executing NoOp(SIP/3000-ca1c, Using CallerID Augusta I.T Tes 3000) in new stack -- Executing Macro(SIP/3000-ca1c, record-enable|3000|OUT) in new stack -- Executing GotoIf(SIP/3000-ca1c, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/3000-ca1c, recordingcheck|20100507-082747|1273235267.398) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20100507-082747|1273235267.398: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/3000-ca1c, No recording needed) in new stack -- Executing Macro(SIP/3000-ca1c, outbound-callerid|3) in new stack -- Executing DBget(SIP/3000-ca1c, USEROUTCID=AMPUSER/3000/outboundcid) in new stack -- DBget: varname=USEROUTCID, family=AMPUSER, key=3000/outboundcid -- DBget: set variable USEROUTCID to -- Executing GotoIf(SIP/3000-ca1c, 1?4) in new stack -- Goto (macro-outbound-callerid,s,4) -- Executing GotoIf(SIP/3000-ca1c, 1?6) in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing NoOp(SIP/3000-ca1c, CallerID set to Augusta I.T Tes 3000) in new stack -- Executing SetGroup(SIP/3000-ca1c, OUT_3) in new stack -- Executing CheckGroup(SIP/3000-ca1c, ) in new stack -- Executing SetVar(SIP/3000-ca1c, DIAL_NUMBER=3643873) in new stack -- Executing SetVar(SIP/3000-ca1c, DIAL_TRUNK=3) in new stack -- Executing AGI(SIP/3000-ca1c, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar(SIP/3000-ca1c, OUTNUM=3643873) in new stack -- Executing Cut(SIP/3000-ca1c, custom=OUT_3|:|1) in new stack -- Executing GotoIf(SIP/3000-ca1c, 0?16) in new stack -- Executing Dial(SIP/3000-ca1c, IAX2/augusta/3643873) in new stack -- Called augusta/3643873 -- Call accepted by 192.168.1.10 (format ulaw) -- Format for call is ulaw -- IAX2/augusta-16384 is making progress passing it to SIP/3000-ca1c -- Hungup 'IAX2/augusta-16384' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/3000-ca1c' in macro 'dialout-trunk' == Spawn extension (from-internal, %233643873, 1) exited non-zero on 'SIP/3000-ca1c' -- Executing Macro(SIP/3000-ca1c, hangupcall) in new stack -- Executing ResetCDR(SIP/3000-ca1c, w) in new stack -- Executing NoCDR(SIP/3000-ca1c, ) in new stack -- Executing Wait(SIP/3000-ca1c, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/3000-ca1c' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3000-ca1c' It stripped the hash and passed the number through the IAX2 trunk. I am just getting a all circuits are busy. Thanks, David On Wed, May 5, 2010 at 6:53 PM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! I set: sip debug peer 3000 (my test extension) and dialed #3643873 Your X-Lite softphone actually calls %233643873 and not #3643873. You would need to check the SIP RFCs in order to find out if Asterisk is behaving correctly here by not decoding %23 as #. In the meanwhile you could try to add the extension %233643873 to your dialplan, or find out if you can configure the way X-Lite handles the # within the dialstring. To: #3643873sip:%233643...@192.168.2.10 mailto:sip%3a%25233643...@192.168.2.10 ... User-Agent: X-Lite release 1104o stamp 56125
Re: [asterisk-users] Getting presence working in 1.6.2
Richard Kenner wrote: I read the wiki and see mention about needing to set call-limit in asterisk 1.4 but that has been depreciated in 1.6 so what is the way it should be done in 1.6? I set callcounter=yes in sip.conf. Thanks that works perfectly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Getting presence working in 1.6.2
In which future release of Asterisk are we (since it is open-source, we theoretically have some control) going to stop renaming and deprecating features? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades Sent: Friday, May 07, 2010 8:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Getting presence working in 1.6.2 Richard Kenner wrote: I read the wiki and see mention about needing to set call-limit in asterisk 1.4 but that has been depreciated in 1.6 so what is the way it should be done in 1.6? I set callcounter=yes in sip.conf. Thanks that works perfectly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting presence working in 1.6.2
On Friday 07 May 2010 08:25:17 Danny Nicholas wrote: In which future release of Asterisk are we (since it is open-source, we theoretically have some control) going to stop renaming and deprecating features? I doubt that will ever happen. In the case of callcounter, that's not a rename, anyway; it's merely a reflection of the fact that people want devicestate, yet limiting calls some some arbitrary number is the legacy interface (and a bad one, at that). We will always reserve the right to make improvements. Sometimes, that means deprecating poor interfaces. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting presence working in 1.6.2
Do you see the creation of a buyer beware repository that host deprecated features (like agentcallbacklogin) that aren't happy for current release but might be desired for backward compatibility? Or is that just a port that we would bring forward ourselves outside of the norm? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, May 07, 2010 8:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Getting presence working in 1.6.2 On Friday 07 May 2010 08:25:17 Danny Nicholas wrote: In which future release of Asterisk are we (since it is open-source, we theoretically have some control) going to stop renaming and deprecating features? I doubt that will ever happen. In the case of callcounter, that's not a rename, anyway; it's merely a reflection of the fact that people want devicestate, yet limiting calls some some arbitrary number is the legacy interface (and a bad one, at that). We will always reserve the right to make improvements. Sometimes, that means deprecating poor interfaces. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting presence working in 1.6.2
On Fri, 2010-05-07 at 08:25 -0500, Danny Nicholas wrote: In which future release of Asterisk are we (since it is open-source, we theoretically have some control) going to stop renaming and deprecating features? It's obviously more complicated that you make it seem with your comment. Let me try to explain the history of this particular change. In earlier versions of Asterisk (1.2, 1.4, 1.6.0 and deprecated but still working in 1.6.1), you had to set the call-limit setting to get Asterisk to keep track of SIP device state. The majority of the people using this call-limit setting set it to an arbitrarily high value (such as 99) so that it didn't really limit the number of concurrent calls, but simply turned on SIP device state tracking. (And, to be honest, it was a whole lot easier to use the GROUP() and GROUP_COUNT() functions in the dialplan to enforce arbitrary call limits.) To make it more clear and less cryptic, we split out the callcounter functionality in sip.conf, so that you could turn on/off the SIP device state tracking without limiting calls, and encouraged people to use the GROUP() and GROUP_COUNT() functions in the dialplan to enforce call limits. Clear as mud? -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Contact header appears incorrect on this invite Asterisk registering with another PBX
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
Your registration looks all wrong. The contact header appears
incorrect on this invite. Please make it read
Contact: sip:{broadsmart_us...@{our_ip}:5060
This is probably the userid or auth user id.
REGISTER sip:{broadsmart_ip} SIP/2.0
Via: SIP/2.0/UDP {our_ip}:5060;branch=z9hG4bK1e85dd83;rport
Max-Forwards: 70
From: sip:{broadsmart_us...@{broadsmart_ip};tag=as3bafb590
To: sip:{broadsmart_us...@{broadsmart_ip}
Call-ID: 13545ba119fb96b707e90636720df...@127.0.0.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX 1.6.2.5
Expires: 120
Contact: sip:s...@{our_ip}
Content-Length: 0
Please change expires to what we are configured which is 3600 seconds.
I'm not sure what it is that may be causing the Contact to show up as s.
Here are the associated configs.
sip.conf
[general]
register = {broadsmart_user}:{broadsmart_passwo...@{broadsmart_ip}
[broadsmart]
host={broadsmart_ip}
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser={broadsmart_user}
secret={broadsmart_password}
fromdomain=broadsmart.net
quality=3600
canreinvite=no
Sorry for the long request. Admittedly I'm lost.
--
Mike A. Leonetti
As warm as green tea
--
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Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
--
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Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
--
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Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
No. The server you register with can often be different to the one you
pass calls to so keeping them completely separate makes a lot of sense.
You can put the authentication information in the dial command itself
but that is generally not a good idea because it can expose the username
and password to other applications which integrate into asterisk or when
viewing the asterisk console.
--
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Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
No. The server you register with can often be different to the one you
pass calls to so keeping them completely separate makes a lot of sense.
You can put the authentication information in the dial command itself
but that is generally not a good idea because it can expose the username
and password to other applications which integrate into asterisk or when
viewing the asterisk console.
So then where is my mistake? The credentials in broadsmart look like
the same from whats being registered.
--
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[asterisk-users] Asterisk Bible?
Hi Folks, Is there a generally accepted Asterisk bible for current versions? I poked around the forums and there didn't seem to be a real consensus, and there are lots of options out there. I need something that focuses on Asterisk dialplans and config files, not a linux primer. I'm looking for dead-tree rather than online documentation. Thanks, Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid =
10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
No. The server you register with can often be different to the one you
pass calls to so keeping them completely separate makes a lot of sense.
You can put the authentication information in the dial command itself
but that is generally not a good idea because it can expose the username
and password to other applications which integrate into asterisk or when
viewing the asterisk console.
So then where is my mistake? The credentials in broadsmart look like
the same from whats being registered.
I cant say but just made you aware that both are separate so the
password may be wrong in one place. It would be best to do a sip debug
and that may help diagnose the problem.
I am off now so wont be back until after the weekend so hopefully
someone else will help furthur.
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Re: [asterisk-users] Asterisk Bible?
I personally own (or have owned) about six different asterisk books, and this one was far the most instrumental. Asterisk: The Future of Telephony, 2nd Edition, dead tree edition http://www.amazon.com/Asterisk-Telephony-Jim-Van-Meggelen/dp/0596510489/ref=sr_1_1 -Karl - Original Message - From: Tim Densmore tdensm...@tarpit.cybermesa.com To: asterisk-users@lists.digium.com Sent: Friday, May 07, 2010 11:37 AM Subject: [asterisk-users] Asterisk Bible? Hi Folks, Is there a generally accepted Asterisk bible for current versions? I poked around the forums and there didn't seem to be a real consensus, and there are lots of options out there. I need something that focuses on Asterisk dialplans and config files, not a linux primer. I'm looking for dead-tree rather than online documentation. Thanks, Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
On 05/07/10 12:40, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register
to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid =
10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
No. The server you register with can often be different to the one you
pass calls to so keeping them completely separate makes a lot of sense.
You can put the authentication information in the dial command itself
but that is generally not a good idea because it can expose the username
and password to other applications which integrate into asterisk or when
viewing the asterisk console.
So then where is my mistake? The credentials in broadsmart look like
the same from whats being registered.
I cant say but just made you aware that both are separate so the
password may be wrong in one place. It would be best to do a sip debug
and that may help diagnose the problem.
I am off now so wont be back until after the weekend so hopefully
someone else will help furthur.
It turns out that it's actually on the registration end. I see that too:
[May 7 13:02:14] NOTICE[10402]: chan_sip.c:11461 sip_reregister:--
Re-registration for {broadsmart_passwo...@{broadsmart_ip}
REGISTER 12 headers, 0
lines
Reliably Transmitting (no NAT) to
{broadsmart_ip}:5060:
REGISTER sip:{broadsmart_ip}
SIP/2.0
Via: SIP/2.0/UDP
{asterisk_ip}:5060;branch=z9hG4bK6df043c0;rport
Max-Forwards:
70
From:
sip:{broadsmart_passwo...@{broadsmart_ip};tag=as59ede08c
To:
sip:{broadsmart_passwo...@{broadsmart_ip}
Call-ID:
4fd754b9115b2e1c2c17ce6d1f24b...@127.0.0.1
CSeq: 104
REGISTER
User-Agent: Asterisk PBX
1.6.2.5
Authorization: Digest username={broadsmart_password},
realm=Registered_Subscribers, algorithm=MD5, uri=sip:broadsmart.net,
nonce=c022714eff5d7016afe930e9390392a3,
response=2e14289556acb0bf2657504c9147b6c1,
opaque=e5677a6b
Expires:
3600
Contact:
sip:s...@{asterisk_ip}
Content-Length: 0
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Re: [asterisk-users] Asterisk Bible?
When something happens this will be good... http://asteriskcookbook.com/wiki/index.php/Main_Page ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Fri, May 7, 2010 at 1:09 PM, Karl Fife karlf...@gmail.com wrote: I personally own (or have owned) about six different asterisk books, and this one was far the most instrumental. Asterisk: The Future of Telephony, 2nd Edition, dead tree edition http://www.amazon.com/Asterisk-Telephony-Jim-Van-Meggelen/dp/0596510489/ref=sr_1_1 -Karl - Original Message - From: Tim Densmore tdensm...@tarpit.cybermesa.com To: asterisk-users@lists.digium.com Sent: Friday, May 07, 2010 11:37 AM Subject: [asterisk-users] Asterisk Bible? Hi Folks, Is there a generally accepted Asterisk bible for current versions? I poked around the forums and there didn't seem to be a real consensus, and there are lots of options out there. I need something that focuses on Asterisk dialplans and config files, not a linux primer. I'm looking for dead-tree rather than online documentation. Thanks, Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Execute AGI, Then Continue
Hi all, I'm running Asterisk 1.6.2.7 using the following pseudo-dialplan (not actual dialplan, because of complexity): [something] exten = s,1,Answer() exten = s,n,AGI(blah,arg1,arg2) exten = s,n,Playback(blah) exten = s,n,DoMoreStuff() exten = s,n,Hangup() What I'd like to do, is have Asterisk launch my AGI script and continue executing dialplan without waiting for the AGI to finish executing. I'm aware that I can do this manually in my AGI by forking, but I'd like to avoid doing it that way if possible. I remember reading something a long time ago saying that there was a way to do this, but I can't seem to find that documentation again. Am I crazy, or is this possible to do without modifying my code? Thanks for all help. -Randall -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Contact header appears incorrect on this invite Asterisk registering with another PBX
On 05/07/10 12:40, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register
to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid =
10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
No. The server you register with can often be different to the one you
pass calls to so keeping them completely separate makes a lot of sense.
You can put the authentication information in the dial command itself
but that is generally not a good idea because it can expose the username
and password to other applications which integrate into asterisk or when
viewing the asterisk console.
So then where is my mistake? The credentials in broadsmart look like
the same from whats being registered.
I cant say but just made you aware that both are separate so the
password may be wrong in one place. It would be best to do a sip debug
and that may help diagnose the problem.
I am off now so wont be back until after the weekend so hopefully
someone else will help furthur.
I see what it is. It was the contact extension value that wasn't set.
It defaults to s. Adding a / and putting that contact extension
afterwards fixed the problem. The phones still aren't working, but
thanks for all of the help.
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
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Re: [asterisk-users] Execute AGI, Then Continue
Why not do System(blah) instead of AGI(blah) unless there is some AGI specific item you need. To quote Steve Edwards, Do your AGI in C so it will run 100 times faster and don't worry about when it comes back. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Randall Degges Sent: Friday, May 07, 2010 12:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Execute AGI, Then Continue Hi all, I'm running Asterisk 1.6.2.7 using the following pseudo-dialplan (not actual dialplan, because of complexity): [something] exten = s,1,Answer() exten = s,n,AGI(blah,arg1,arg2) exten = s,n,Playback(blah) exten = s,n,DoMoreStuff() exten = s,n,Hangup() What I'd like to do, is have Asterisk launch my AGI script and continue executing dialplan without waiting for the AGI to finish executing. I'm aware that I can do this manually in my AGI by forking, but I'd like to avoid doing it that way if possible. I remember reading something a long time ago saying that there was a way to do this, but I can't seem to find that documentation again. Am I crazy, or is this possible to do without modifying my code? Thanks for all help. -Randall -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] What is billsec in CDR?
I posted a topic about the billsec.
Asterisk around to the lesser int sec.
Exemple: If the call duration is 15.9 seconds, billsec (ANSWEREDTIME) will
be 15
I could change somethings in the source to have a correct calculation.
Exemple: : If the call duration is 15.9 seconds, then billsec (ANSWEREDTIME)
will be 16.
I changes round to second nearest int second. If duration=15.4 billsec=15.
If duration=15.6 billsec=16.
François Berganz
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Jian Gao
Envoyé : jeudi 6 mai 2010 19:34
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] What is billsec in CDR?
Philipp von Klitzing wrote:
Hi!
apps like playback do an implicit answer and this fires up the
billsec counter.
OK, here is my dialplan:
exten = _011X.,n,Playback(this-call-will-end-in)
exten =
_011X.,n,Dial(SIP/${ext...@${ldtrunk1},60,L(${ms}:3))
Is there any way that Asterisk will record the correct billsec? Or,
is there a different approach?
Place a ResetCDR() after your Playback() statement and before Dial().
Philipp
ResetCDR() works!
Thank you very much!
--
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IT Technician
SJ Geophysics Ltd. http://www.sjgeophysics.com jian@sjgeophysics.com
mailto:jian@sjgeophysics.com
Tel: (604)582-1100
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Re: [asterisk-users] Questions About Fax for Asterisk
On Thu, May 6, 2010 at 3:11 PM, Steve Totaro stot...@totarotechnologies.com wrote: Yes, I purchased licenses for Fax for Asterisk and yes I called tech support and had the WORST experience I have ever had with any technical support call. I am running Asterisk 1.6.2.6 and: FAX For Asterisk Components: Applications: 1.6.2.0_1.2.0 voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32) The guy was arrogant and absolutely a jerk and I don't like to call people names, but call it as I see it. This has not been my experience the five or six times I have had to call Digium over the years, but it has been many years since my last call so I have no idea what the general support staff is like. I could not get any questions answered by the tech that took hours to call me back to tell me to read the readme. That would be all well and good if I didn't pay money. He could not explain Digium's math as far as faxing and failed to offer to get back to me with any kind of answer. Maybe someone on the list can make sense of this Enron style of accounting: voipgw01*CLI fax show stats voipgw01*CLI FAX Statistics: --- Current Sessions : 1 Transmit Attempts : 0 Receive Attempts : 336 Completed FAXes : 320 Failed FAXes : 57 Digium G.711 Licensed Channels : 4 Max Concurrent : 1 Success : 0 Switched to T.38 : 0 Canceled : 0 No FAX : 1 Partial : 0 Negotiation Failed : 0 Train Failure : 3 Protocol Error : 0 IO Partial : 0 IO Fail : 0 voipgw01*CLI Digium T.38 Licensed Channels : 4 Max Concurrent : 4 Success : 175 Canceled : 0 No FAX : 6 Partial : 19 Negotiation Failed : 0 Train Failure : 83 Protocol Error : 33 IO Partial : 0 IO Fail : 0 Thanks, Steve Totaro wow definitely the acccounting engine is broken ... I can only make sense of this Receive Attempts : 336 Completed FAXes : 320 Failed FAXes : 57 1) your receive app was called 336 times but the fax hanged up before negotiating 2) you had 320 of this completed (partially or fully) 3) but 57 out of 320 failed to transmit entirely 57/320=17.8% which is too high for a commercial product IHMO Martin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple SIP lines.
All:
Still experimenting with the asterisk server for the company.
My local phone company has given me two sip numbers to experiment with,
say 444-456-1234 444-456-5678
Calling in and out works, and I've spread a couple of the phones out
with some co-workers.
My question is this: Do I have to define multiple sip lines in either
the sip.conf or the extensions.conf?
Now when I dial out, I just use
exten = _9.,1,DIAL(SIP/${EXTEN:1...@xx.tracfone.net).
How does it know which sip channel to use?
Hope that is clear.
Thanks for all the help.
Eddie Mikell
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Re: [asterisk-users] Getting presence working in 1.6.2
On Fri, May 7, 2010 at 3:41 AM, Jared Smith jsm...@digium.com wrote: To make it more clear and less cryptic, we split out the callcounter functionality in sip.conf, so that you could turn on/off the SIP device state tracking without limiting calls, and encouraged people to use the GROUP() and GROUP_COUNT() functions in the dialplan to enforce call limits. But why 'callcounter', it is frustratingly close 'call-limit' and there is no possible way to use logic to determine what it does. If a change was to be made, why not use 'devicestatetracking=yes'? -matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Getting presence working in 1.6.2
On Friday 07 May 2010 13:59:23 Matt Darnell wrote: On Fri, May 7, 2010 at 3:41 AM, Jared Smith jsm...@digium.com wrote: To make it more clear and less cryptic, we split out the callcounter functionality in sip.conf, so that you could turn on/off the SIP device state tracking without limiting calls, and encouraged people to use the GROUP() and GROUP_COUNT() functions in the dialplan to enforce call limits. But why 'callcounter', it is frustratingly close 'call-limit' and there is no possible way to use logic to determine what it does. If a change was to be made, why not use 'devicestatetracking=yes'? As it's now in three releases, it's rather late to be changing it, although we could add an additional alias. You should probably be watching the commits list and send an email to the -dev list anytime you see something that you think could be better named. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP REGISTER header not containing Allow-Events or Allow
The SIP trunking service that I am trying to set up keeps saying that my
registration from Asterisk is invalid.
Asterisk registration:
REGISTER sip:{registration_ip} SIP/2.0
Via: SIP/2.0/UDP {asterisk_ip}:5060;branch=z9hG4bK5c2eb10c;rport
Max-Forwards: 70
From: sip:{registration_us...@{registration_ip};tag=as5579cc0c
To: sip:{registration_us...@{registration_ip}
Call-ID: 651194bd76e02f4d0126373c51568...@127.0.0.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 1.6.2.5
Authorization: Digest username={registration_user},
realm=Registered_Subscribers, algorithm=MD5, uri=sip:broadsmart.net,
nonce=b47c87b5e93ba420a0cf25162fa29794,
response=98e4d21ca0f75497d7fb12a8a4914bcb, opaque=5adc3dd2
Expires: 3600
Contact: sip:{registration_us...@{asterisk_ip}
Content-Length: 0
Expected registration:
REGISTER sip:broadsmart.net SIP/2.0
Via: SIP/2.0/UDP
208.73.25.70:5060;branch=z9hG4bK-d87543-826b1b62b62ac91d-1--d87543-;rport
Record-Route: sip:2135997...@208.73.25.70;lr
From: 2135997816 sip:2135997...@broadsmart.net;tag=e944c233
To: 2135997816 sip:2135997...@broadsmart.net
Call-ID: e0576109f9699...@dgfjd3mxlmludc5uyxrlbgnvbw0uy29t
CSeq: 1 REGISTER
Contact: sip:2135997...@208.73.25.70:5060;rinstance=92c0558ad60f5de4
Max-forwards: 70
Expires: 3600
Supported: eventlist
User-agent: CounterPath eyeBeam release 3014w stamp 26275
*Allow-Events: BroadWorksSubscriberData
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO*
Content-Length: 0
They are saying that the Asterisk registration doesn't have an
Allow-Events and an Allow in the header. Would this cause any
problems and can this be set in Asterisk to send those in the header?
Thanks.
--
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[asterisk-users] voipmonitor.org
Hi, checkout new open source voipmonitor.org SIP packet sniffer. I've developed it for my telco company and I've decided to share it. Testing and contributions are welcome! VoIPmonitor is open source live network packet sniffer which analyze SIP and RTP protocol. It can run as daemon or analyzes already captured pcap files. For each detected VoIP call voipmonitor calculates statistics about loss, burstiness, latency and predicts MOS (Meaning Opinion Score) according to ITU-T G.107 E-model. These statistics are saved to MySQL database and each call is saved as pcap dump. Web PHP application (it is not part of open source sniffer) filters data from database and graphs latency and loss distribution. Voipmonitor also detects improperly terminated calls when BYE or OK was not seen. To accuratly transform latency to loss packets, voipmonitor simulates fixed and adaptive jitterbuffer. Key features Fast C++ SIP/RTP packet analyzer Predicts MOS-LQE score according to ITU-T G.107 E-model Detailed delay/loss statistics stored to MySQL Each call is saved as standalone pcap file Jitterbuffer simulator based on asterisk (fixed/adaptive) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] voipmonitor.org
Martin- checkout new open source voipmonitor.org SIP packet sniffer. I've developed it for my telco company and I've decided to share it. Testing and contributions are welcome! VoIPmonitor is open source live network packet sniffer which analyze SIP and RTP protocol. It can run as daemon or analyzes already captured pcap files. For each detected VoIP call voipmonitor calculates statistics about loss, burstiness, latency and predicts MOS (Meaning Opinion Score) according to ITU-T G.107 E-model. These statistics are saved to MySQL database and each call is saved as pcap dump. Web PHP application (it is not part of open source sniffer) filters data from database and graphs latency and loss distribution. Voipmonitor also detects improperly terminated calls when BYE or OK was not seen. To accuratly transform latency to loss packets, voipmonitor simulates fixed and adaptive jitterbuffer. How many channels can it handle simultaneously? How does it do MOS prediction if low bitrate codecs are being used (G729, AMR, etc)? Thanks. -Jeff Key features Fast C++ SIP/RTP packet analyzer Predicts MOS-LQE score according to ITU-T G.107 E-model Detailed delay/loss statistics stored to MySQL Each call is saved as standalone pcap file Jitterbuffer simulator based on asterisk (fixed/adaptive) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] DAHDI astribank Channel Unavailable
Hi guys, This has been bugging me for days and I can't figure out what's happening. Using the same channel on DAHDI, I can't make a call after previous one being hung up by remote side. However, it works like it should if my end hangs up the call. To make things interesting, DAHDI, or astribank rather, seems to reset itself after receiving an incoming call on that DAHDI channel. I looked through asterisk as well as kernel logs, didn't find much. Here's some CLI outputs. Does anybody have some suggestions what might have gone wrong? asterisk says: -- Executing [311xxx...@3001-outbound:1] Dial(SIP/3001-0a02, DAHDI/31/1xx) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/3001-0a02' status is 'CHANUNAVAIL' dahdi show channel 31 says: alpha-1*CLI dahdi show channel 31 Channel: 31 File Descriptor: 23 Span: 4 Extension: Dialing: no Context: from-pstn Caller ID: Calling TON: 0 Caller ID name: Mailbox: none Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: yes Busy Count: 5 Busy Pattern: 0,0 TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 128 taps currently OFF Wait for dialtone: 0ms Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook Kelvin Chan NOTICE: This communication is intended only for the use of the person or entity named above and may contain information that is confidential or legally privileged. If you are not the intended recipient named above or a person responsible for delivering messages or communication to the intended recipient, you are hereby notified that any use, distribution, or copying of this communication or any of the information contained in it is strictly prohibited. If you have this communication in error, please notify me immediately by telephone and then destroy or delete this communication, or return it to me by mail if requested. Thank you for your attention and cooperation. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Multiple SIP lines.
I think it is typical to have some limited number of outbound channels to your
SIP provider. You send all calls, up to your limit, to the same place. The
phone numbers your provider gave you are used to route inbound calls to your
asterisk box. You will typically have some limited number of inbound channels.
All people could call the same number, again controlled by the number of
channels your provider allows. A reason to have multiple inbound (DID) numbers
is so you can route each number to a specific dialplan extension. You might
route one number to the CEO of the company and the other to a voice tree that
allows the caller to specify the person's extension they want to talk with.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 7, 2010, at 11:17 AM, Eddie Mikell wrote:
All:
Still experimenting with the asterisk server for the company.
My local phone company has given me two sip numbers to experiment with,
say 444-456-1234 444-456-5678
Calling in and out works, and I've spread a couple of the phones out
with some co-workers.
My question is this: Do I have to define multiple sip lines in either
the sip.conf or the extensions.conf?
Now when I dial out, I just use
exten = _9.,1,DIAL(SIP/${EXTEN:1...@xx.tracfone.net).
How does it know which sip channel to use?
Hope that is clear.
Thanks for all the help.
Eddie Mikell
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[asterisk-users] text
Does anyone know how to send a text message from Asterisk? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Questions About Fax for Asterisk
On Fri, May 7, 2010 at 2:01 PM, Martin asteriskl...@callthem.info wrote: On Thu, May 6, 2010 at 3:11 PM, Steve Totaro stot...@totarotechnologies.com wrote: Yes, I purchased licenses for Fax for Asterisk and yes I called tech support and had the WORST experience I have ever had with any technical support call. I am running Asterisk 1.6.2.6 and: FAX For Asterisk Components: Applications: 1.6.2.0_1.2.0 voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32) The guy was arrogant and absolutely a jerk and I don't like to call people names, but call it as I see it. This has not been my experience the five or six times I have had to call Digium over the years, but it has been many years since my last call so I have no idea what the general support staff is like. I could not get any questions answered by the tech that took hours to call me back to tell me to read the readme. That would be all well and good if I didn't pay money. He could not explain Digium's math as far as faxing and failed to offer to get back to me with any kind of answer. Maybe someone on the list can make sense of this Enron style of accounting: voipgw01*CLI fax show stats voipgw01*CLI FAX Statistics: --- Current Sessions : 1 Transmit Attempts: 0 Receive Attempts : 336 Completed FAXes : 320 Failed FAXes : 57 Digium G.711 Licensed Channels: 4 Max Concurrent : 1 Success : 0 Switched to T.38 : 0 Canceled : 0 No FAX : 1 Partial : 0 Negotiation Failed : 0 Train Failure: 3 Protocol Error : 0 IO Partial : 0 IO Fail : 0 voipgw01*CLI Digium T.38 Licensed Channels: 4 Max Concurrent : 4 Success : 175 Canceled : 0 No FAX : 6 Partial : 19 Negotiation Failed : 0 Train Failure: 83 Protocol Error : 33 IO Partial : 0 IO Fail : 0 Thanks, Steve Totaro wow definitely the acccounting engine is broken ... I can only make sense of this Receive Attempts : 336 Completed FAXes : 320 Failed FAXes : 57 1) your receive app was called 336 times but the fax hanged up before negotiating 2) you had 320 of this completed (partially or fully) 3) but 57 out of 320 failed to transmit entirely 57/320=17.8% which is too high for a commercial product IHMO Martin Considering that this is a direct cross connect from Leve3's cage to my my cage in the same DC at an Equinix facility, 100Mb DIA w/EIPT VoIP service, I would expect nearly 100% success. Considering the circuit was just turned up and there is no data except Level3's phone traffic. They are our carrier, RespOrg, origination and termination, no 3rd parties, all on net. I could understand if it was a peaked out DIA circuit to some cut rate VoIP provider, but not under perfect circumstances. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] text
On Fri, 7 May 2010, Thomas Perron wrote: Does anyone know how to send a text message from Asterisk? Carrier specific, but how about: system(echo foo | mail -s bar 551...@txt.att.net) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions About Fax for Asterisk
On 05/08/2010 08:15 AM, Steve Totaro wrote: On Fri, May 7, 2010 at 2:01 PM, Martin asteriskl...@callthem.info mailto:asteriskl...@callthem.info wrote: On Thu, May 6, 2010 at 3:11 PM, Steve Totaro stot...@totarotechnologies.com mailto:stot...@totarotechnologies.com wrote: Yes, I purchased licenses for Fax for Asterisk and yes I called tech support and had the WORST experience I have ever had with any technical support call. I am running Asterisk 1.6.2.6 and: FAX For Asterisk Components: Applications: 1.6.2.0_1.2.0 voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32) The guy was arrogant and absolutely a jerk and I don't like to call people names, but call it as I see it. This has not been my experience the five or six times I have had to call Digium over the years, but it has been many years since my last call so I have no idea what the general support staff is like. I could not get any questions answered by the tech that took hours to call me back to tell me to read the readme. That would be all well and good if I didn't pay money. He could not explain Digium's math as far as faxing and failed to offer to get back to me with any kind of answer. Maybe someone on the list can make sense of this Enron style of accounting: voipgw01*CLI fax show stats voipgw01*CLI FAX Statistics: --- Current Sessions : 1 Transmit Attempts: 0 Receive Attempts : 336 Completed FAXes : 320 Failed FAXes : 57 Digium G.711 Licensed Channels: 4 Max Concurrent : 1 Success : 0 Switched to T.38 : 0 Canceled : 0 No FAX : 1 Partial : 0 Negotiation Failed : 0 Train Failure: 3 Protocol Error : 0 IO Partial : 0 IO Fail : 0 voipgw01*CLI Digium T.38 Licensed Channels: 4 Max Concurrent : 4 Success : 175 Canceled : 0 No FAX : 6 Partial : 19 Negotiation Failed : 0 Train Failure: 83 Protocol Error : 33 IO Partial : 0 IO Fail : 0 Thanks, Steve Totaro wow definitely the acccounting engine is broken ... I can only make sense of this Receive Attempts : 336 Completed FAXes : 320 Failed FAXes : 57 1) your receive app was called 336 times but the fax hanged up before negotiating 2) you had 320 of this completed (partially or fully) 3) but 57 out of 320 failed to transmit entirely 57/320=17.8% which is too high for a commercial product IHMO Martin Considering that this is a direct cross connect from Leve3's cage to my my cage in the same DC at an Equinix facility, 100Mb DIA w/EIPT VoIP service, I would expect nearly 100% success. Considering the circuit was just turned up and there is no data except Level3's phone traffic. They are our carrier, RespOrg, origination and termination, no 3rd parties, all on net. I could understand if it was a peaked out DIA circuit to some cut rate VoIP provider, but not under perfect circumstances. Thanks, Steve Totaro Were these all test calls made from a well defined source? It takes *two* correctly working FAX terminals to make a successful call. Its easy to get a high failure rate for silly reasons. In volume testing of spandsp and iaxmodem we had times where a high percentage of calls failed, which turned out to be just one rouge machine calling over and over again trying to achieve success. On the other hand, failures between known good FAX terminals should be far below 1%. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] text
thanks do i need to have an smtp server somewhere. i tried directly from my dialplan but no joy! i know you know that i am not a star with this but any help would be cool here is my config: exten = 600,1,Answer() exten = 600,n,Wait(1) exten = 600,n,system(echo foo | mail -s bar 2224441...@txt.att.net) On Fri, May 7, 2010 at 8:32 PM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 7 May 2010, Thomas Perron wrote: Does anyone know how to send a text message from Asterisk? Carrier specific, but how about: system(echo foo | mail -s bar 551...@txt.att.net) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
