[asterisk-users] Connecting Multiple Asterisk instances getting Unable to create channel of type 'SIP'
Hi, I'm trying to connect two asterisk instances using the method described here.. http://ofps.oreilly.com/titles/9781449332426/asterisk-OutsideConn.html under the section Connecting two Asterisk systems together with SIP I have an user named venu in serverA and vijay in serverB the serverA ip is 192.168.0.35 serverB is 192.168.0.36 Here are the details of the config files (extension sip): http://paste.kde.org/737888 When i make a call to extension 998 in using user as venu, here is the output i get.. http://paste.kde.org/737894 The problem is that, I'm getting the *Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)* but I want to make a call to vijay.. can anyone please let me know where I am going wrong? I have the same error when I try to make a call from sip client to a analog phone in a single server asterisk setup... :-\ I'm running Asterisk 11.3 on Ubuntu 12.04 on a KVM virtualized instance.. -- _ -- Bandwidth and Colocation Provided by http://www.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] GotoIf DIALSTATUS - not working
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Sunday, 5 May 2013 5:33 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] GotoIf DIALSTATUS - not working
What am I doing wrong?
Goif dialstatus: busy CONGESTION not working.
exten = _7NXX,1,Dial(SIP/7780${EXTEN:1}@pstn-5665,60,tr)
exten = _7NXX,n,GotoIf($[$[${DIALSTATUS} = BUSY] |
$[${DIALSTATUS} = CONGESTION]]?line2)
exten = _7NXX,n(line2),Dial(SIP/9780${EXTEN:1}@pstn-1270,60,tr)
exten = _7NXX,n,Hangup()
When I try to call another number (7780476444) on a different
line it supposed to jump to (line2) on busy (and dial
Dial(SIP/9780${EXTEN:1}@pstn-1270) but instead the call hangs up.
-- Called SIP/7780476@pstn-5665
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [7476@internal:5]
Hangup(SIP/11-015b, ) in new stack
--
Joseph
--
I'd suggest a line to print the DIALSTATUS as below.
I get CHANUNAVAIL, thus hangs up.
exten = _7NXX,1,Dial(SIP/7780${EXTEN:1}@pstn-5665,60,tr)
exten = _7NXX,n,NoOp(DialStatus=${DIALSTATUS})
exten = _7NXX,n,GotoIf($[$[${DIALSTATUS} = BUSY] |
$[${DIALSTATUS} = CONGESTION]]?line2)
exten = _7NXX,n,Hangup()
exten = _7NXX,n(line2),Dial(SIP/9780${EXTEN:1}@pstn-1270,60,tr)
exten = _7NXX,n,NoOp(DialStatus=${DIALSTATUS})
exten = _7NXX,n,Hangup()
Alec
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Re: [asterisk-users] Connecting Multiple Asterisk instances getting Unable to create channel of type 'SIP'
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sandeep Raju Sent: Sunday, 5 May 2013 8:34 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Connecting Multiple Asterisk instances getting Unable to create channel of type 'SIP' snip When i make a call to extension 998 in using user as venu, here is the output i get.. http://paste.kde.org/737894 The problem is that, I'm getting the Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) but I want to make a call to vijay.. can anyone please let me know where I am going wrong? The clue is 21.-- Executing [999@incoming:2] Dial(SIP/serverA-0004, SIP/vijay@serverB) in new stack 24. getaddrinfo(serverB, (null), ...): Name or service not known 25. No such host: serverB I believe extension 999 in server B is wrong. It should be; # extensions.conf in serverB [incoming] exten = 999,1,Answer() exten = 999,n,Dial(SIP/vijay) exten = 999,n,HangUp() Alec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Multiple Asterisk instances getting Unable to create channel of type 'SIP'
@Alec, Thanks.. That was the error.. got it working now.. :) On Sun, May 5, 2013 at 2:34 PM, Alec Davis siva...@paradise.net.nz wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sandeep Raju Sent: Sunday, 5 May 2013 8:34 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Connecting Multiple Asterisk instances getting Unable to create channel of type 'SIP' snip When i make a call to extension 998 in using user as venu, here is the output i get.. http://paste.kde.org/737894 The problem is that, I'm getting the Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) but I want to make a call to vijay.. can anyone please let me know where I am going wrong? The clue is 21.-- Executing [999@incoming:2] Dial(SIP/serverA-0004, SIP/vijay@serverB) in new stack 24. getaddrinfo(serverB, (null), ...): Name or service not known 25. No such host: serverB I believe extension 999 in server B is wrong. It should be; # extensions.conf in serverB [incoming] exten = 999,1,Answer() exten = 999,n,Dial(SIP/vijay) exten = 999,n,HangUp() Alec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Multiple Asterisk instances getting Unable to create channel of type 'SIP'
@Alec, Now I can dial user vijay but the call gets cut after a few seconds and i get this error in the serverA's console.. http://paste.kde.org/737924 PS: recolgo is the hostname of the system from which I am initialting the call (using a sip client) Thanks On Sun, May 5, 2013 at 2:41 PM, Sandeep Raju sandeepr...@practo.com wrote: @Alec, Thanks.. That was the error.. got it working now.. :) On Sun, May 5, 2013 at 2:34 PM, Alec Davis siva...@paradise.net.nzwrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sandeep Raju Sent: Sunday, 5 May 2013 8:34 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Connecting Multiple Asterisk instances getting Unable to create channel of type 'SIP' snip When i make a call to extension 998 in using user as venu, here is the output i get.. http://paste.kde.org/737894 The problem is that, I'm getting the Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) but I want to make a call to vijay.. can anyone please let me know where I am going wrong? The clue is 21.-- Executing [999@incoming:2] Dial(SIP/serverA-0004, SIP/vijay@serverB) in new stack 24. getaddrinfo(serverB, (null), ...): Name or service not known 25. No such host: serverB I believe extension 999 in server B is wrong. It should be; # extensions.conf in serverB [incoming] exten = 999,1,Answer() exten = 999,n,Dial(SIP/vijay) exten = 999,n,HangUp() Alec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Multiple Asterisk instances getting Unable to create channel of type 'SIP'
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sandeep Raju Sent: Sunday, 5 May 2013 9:19 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Connecting Multiple Asterisk instances getting Unable to create channel of type 'SIP' @Alec, Now I can dial user vijay but the call gets cut after a few seconds and i get this error in the serverA's console.. http://paste.kde.org/737924 From http://paste.kde.org/737924 SecurityEvent=ChallengeSent,EventTV=1367741794-435078,Severity=Informat ional,Service=SIP,EventVersion=1,AccountID=sip:venu@192.168.0.35,Sess ionID=0x337bf68,LocalAddress=IPV4/UDP/10.10.1.3/5060,RemoteAddress=IPV4 /UDP/192.168.1.90/5060,Challenge=41cdcd16 ^^^ The other networks confuse me, and perhaps asterisk. Perhaps serverA:sip.conf udpbindaddr=192.168.0.35 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) serverB:sip.conf udpbindaddr=192.168.0.36 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) Alec -- _ -- Bandwidth and Colocation Provided by http://www.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] debug strategy for one-way audio calls
On 04.05.2013, at 20:20, Olivier oza_4...@yahoo.fr wrote: Le 2 mai 2013 13:23, Marie Fischer ma...@vtl.ee a écrit : from time to time, we get so-called simplex / one-way audio calls, where one party cannot hear the other. The only thing in common is that is does happen with calls via SIP trunk, not ISDN and not internal calls. Nothing strange in verbose and SIP logs. Could even be some weird intermittent firewall issue I guess. Which audio flow is missing ? Inbound ? I suppose it should be easier to automatically detect missing inbound audio. Not sure about older calls, but outbound was missing the last few times. We use call recording via MixMonitor and the recording has both flows, so I guess rtp debug would have shown both as well. Apart from logging all traffic 24/7 via tcpdump (not really convenient), can you give me some ideas how to debug this kind of issue? Asterisk 1.8.11-cert10 on CentOS 6.3, if that matters. -- marie -- _ -- Bandwidth and Colocation Provided by http://www.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] debug strategy for one-way audio calls
Le 5 mai 2013 12:19, Marie Fischer ma...@vtl.ee a écrit : On 04.05.2013, at 20:20, Olivier oza_4...@yahoo.fr wrote: Le 2 mai 2013 13:23, Marie Fischer ma...@vtl.ee a écrit : from time to time, we get so-called simplex / one-way audio calls, where one party cannot hear the other. The only thing in common is that is does happen with calls via SIP trunk, not ISDN and not internal calls. Nothing strange in verbose and SIP logs. Could even be some weird intermittent firewall issue I guess. Which audio flow is missing ? Inbound ? I suppose it should be easier to automatically detect missing inbound audio. Not sure about older calls, but outbound was missing the last few times. We use call recording via MixMonitor and the recording has both flows, so I guess rtp debug would have shown both as well. Yes, I agree: rtp debug would probably show both flows. So your asterisk box is quite probably sending rtp data to a SIP trunk which do not forward it to the other party. And this does happen from time to time (not on every call), and for a remote party which is independant from both your provider and your own infrastructure, right ? This is very strange. I would try to find conditions with which I get missing outbound audio, 100% of time but I don't have a clue on how to do it successfully. What does your SIP provider say about this ? Have you met this with another SIP provider ? Apart from logging all traffic 24/7 via tcpdump (not really convenient), can you give me some ideas how to debug this kind of issue? Asterisk 1.8.11-cert10 on CentOS 6.3, if that matters. -- marie -- _ -- Bandwidth and Colocation Provided by http://www.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] My new Polycom 450's can't xfer to 4-digit extension
Mike Diehl wrote: Is there something I need to do for the 450 to make this work? As far as I know, all the Polycoms require a digit map that isn't blank. You're digit maps are blank. There are two place you can have digit maps. In the individual phone configs and in the master sip.cfg. A digit map for one of my 550s are below: dialplan.1.digitmap=8[0-9]|0T|1x|4[4-5]|011xxx.T|*7|*xxx|*xx.T|04xxx|700|9,1[2-9]x|9,[2-9]xx|[2-8]xxx I'd suggest looking into your master phone template for the bad digit map if as you say, the sip.cfg doesn't contain one. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] GotoIf DIALSTATUS - not working
On 05/05/13 20:50, Alec Davis wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Sunday, 5 May 2013 5:33 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] GotoIf DIALSTATUS - not working
What am I doing wrong?
Goif dialstatus: busy CONGESTION not working.
exten = _7NXX,1,Dial(SIP/7780${EXTEN:1}@pstn-5665,60,tr)
exten = _7NXX,n,GotoIf($[$[${DIALSTATUS} = BUSY] |
$[${DIALSTATUS} = CONGESTION]]?line2)
exten = _7NXX,n(line2),Dial(SIP/9780${EXTEN:1}@pstn-1270,60,tr)
exten = _7NXX,n,Hangup()
When I try to call another number (7780476444) on a different
line it supposed to jump to (line2) on busy (and dial
Dial(SIP/9780${EXTEN:1}@pstn-1270) but instead the call hangs up.
-- Called SIP/7780476@pstn-5665
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [7476@internal:5]
Hangup(SIP/11-015b, ) in new stack
--
Joseph
--
I'd suggest a line to print the DIALSTATUS as below.
I get CHANUNAVAIL, thus hangs up.
exten = _7NXX,1,Dial(SIP/7780${EXTEN:1}@pstn-5665,60,tr)
exten = _7NXX,n,NoOp(DialStatus=${DIALSTATUS})
exten = _7NXX,n,GotoIf($[$[${DIALSTATUS} = BUSY] |
$[${DIALSTATUS} = CONGESTION]]?line2)
exten = _7NXX,n,Hangup()
exten = _7NXX,n(line2),Dial(SIP/9780${EXTEN:1}@pstn-1270,60,tr)
exten = _7NXX,n,NoOp(DialStatus=${DIALSTATUS})
exten = _7NXX,n,Hangup()
Alec
Thank Alex, it is working. I was modifying the wrong context :-/
--
Joseph
--
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[asterisk-users] Testing 911 call
How to test 911 call? I'm using Audiocodes and it setup to strip the first number but I've never tested the 911 call. I don't want to go live as they might charge me. -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 9971 help
Getting closer... As a heads-up, the files in the link do not include the locale information (gd-sip.jar), but I have tracked down something suitable for that... The phones now get all the files that are essential, but never register with asterisk (there is no network traffic). The phone logs show: 8360 NOT 00:21:14.387613 CVM-ccsip_register_send_msg: Error: cc_cfg_table is null. Googling this seems to suggest that I am not alone here, and that possibly the SIP 1.9 build on the website below is broken, or if it isn't broken, the format of the SEP.cnf.xml file has subtly changed, but its not clear how. Anyone got a 9971 working with 1.9.2-2SR1-9 want to share their SEPxxx.cnf.xml? I'd like to regress to SIP 1.8, as I think that may fix the problem, but so far I haven't been able to locate, ahem, a copy. Cheers Patrick On 4 May 2013 18:56, Patrick Lidstone patr...@lidstone.net wrote: Stoyan Marinov wrote: Checkout http://firewall.cx for cisco downloads Looks promising - a later firmware load, so the file I was looking for was not present, but still hopeful! Many thanks for the tip, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 9971 help
Yes, bad form to follow up on my own post. Anyway, the secret sauce is indeed a correct SEP.cnf.xml, and a kind lister provided a working model for SIP 1.9. I now have the endpoint registered, should be downhill from here. Full write up will indeed follow. No doubt I will be back soon with more q's Cheers Patrick On 5 May 2013 16:17, Patrick Lidstone patr...@lidstone.net wrote: Getting closer... As a heads-up, the files in the link do not include the locale information (gd-sip.jar), but I have tracked down something suitable for that... The phones now get all the files that are essential, but never register with asterisk (there is no network traffic). The phone logs show: 8360 NOT 00:21:14.387613 CVM-ccsip_register_send_msg: Error: cc_cfg_table is null. Googling this seems to suggest that I am not alone here, and that possibly the SIP 1.9 build on the website below is broken, or if it isn't broken, the format of the SEP.cnf.xml file has subtly changed, but its not clear how. Anyone got a 9971 working with 1.9.2-2SR1-9 want to share their SEPxxx.cnf.xml? I'd like to regress to SIP 1.8, as I think that may fix the problem, but so far I haven't been able to locate, ahem, a copy. Cheers Patrick On 4 May 2013 18:56, Patrick Lidstone patr...@lidstone.net wrote: Stoyan Marinov wrote: Checkout http://firewall.cx for cisco downloads Looks promising - a later firmware load, so the file I was looking for was not present, but still hopeful! Many thanks for the tip, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.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] Testing 911 call
Joseph, I have made a quite a few test calls to 911. They don't charge you and they don't get upset. Just let them know right away it is a non-emergency test call, and then let them know who you are and what you need to verify on their information screen. Mark Engelhardt On May 5, 2013, at 11:07 AM, Joseph wrote: How to test 911 call? I'm using Audiocodes and it setup to strip the first number but I've never tested the 911 call. I don't want to go live as they might charge me. -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.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] Testing 911 call
If there is a non-emergency number you can call and let them know you would like to do some test calls. This also allows you to schedule a time for testing when the PSAP is not as busy allowing for real calls to be handled. On Sun, May 5, 2013 at 11:15 AM, Mark Engelhardt ma...@intuitiveengineering.com wrote: Joseph, I have made a quite a few test calls to 911. They don't charge you and they don't get upset. Just let them know right away it is a non-emergency test call, and then let them know who you are and what you need to verify on their information screen. Mark Engelhardt On May 5, 2013, at 11:07 AM, Joseph wrote: How to test 911 call? I'm using Audiocodes and it setup to strip the first number but I've never tested the 911 call. I don't want to go live as they might charge me. -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.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] Testing 911 call
I actually work in a 911 center. Please do not dial blindly to do a test call. Please call the non-emergency dispatch number, ask if it would be ok to make one or two test calls. If they give you the ok, please complete those calls as quickly as possible as conditions change in an instant. If they give you permission at 9am, don't wait until 5pm to do the test. Further, most 911/PSAP centers are not busy after 10pm local time, please consider doing your testing in the overnight hours. Again, please check with your local 911/PSAP to confirm when their peak times are and try to avoid them. Here is a good script to read when speaking to the 911 call taker: Hello my name is (your name) with (company name). We are performing a test 911 call and would like to confirm some information if you have a moment. (if they answer go ahead, continue with the script. If they advise now is not a good time, say thank you and disconnect. In a 30 to 60 minutes call the non-emergency number and ask if you may make another test call) (continuing the script) Can you please confirm the address that shows up on your phone system please? (wait and confirm the info) Great thank you. If you can, please tell me the number you show we are calling from? (wait and confirm) Can you confirm for me which 911/PSAP center we have reached? ( wait and confirm this is the proper 911/PSAP you need) (if this is the first of several calls:) Thank you very much for your time, we will be making (xx number) of calls in the next few minutes. (if this is the end of your testing:) Thank you very much for your time, this concludes our testing. Good luck with your phone testing. Regards, James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 On Sun, May 5, 2013 at 8:00 PM, Dale Noll dn...@wi.rr.com wrote: If there is a non-emergency number you can call and let them know you would like to do some test calls. This also allows you to schedule a time for testing when the PSAP is not as busy allowing for real calls to be handled. On Sun, May 5, 2013 at 11:15 AM, Mark Engelhardt ma...@intuitiveengineering.com wrote: Joseph, I have made a quite a few test calls to 911. They don't charge you and they don't get upset. Just let them know right away it is a non-emergency test call, and then let them know who you are and what you need to verify on their information screen. Mark Engelhardt On May 5, 2013, at 11:07 AM, Joseph wrote: How to test 911 call? I'm using Audiocodes and it setup to strip the first number but I've never tested the 911 call. I don't want to go live as they might charge me. -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.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] Joining an astablished call
Hi, I don't know how to call this functionality, but what I want to do is join an already established communication between PSTN---FXS_connected_phone using my SIP phone (I have an asterisk v11 with digium TDM400P at home) Is it possible? What I don't want is using the conference sound and menu It's just a normal call between to channels that I have to join for few minutes. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Joining an astablished call
On 05/05/2013 08:34 PM, neo haux wrote: I don't know how to call this functionality, but what I want to do is join an already established communication between PSTN---FXS_connected_phone using my SIP phone (I have an asterisk v11 with digium TDM400P at home) I had this set up once upon a time. It took quite a bit of dialplan hackery, but the basic idea is to create a conference bridge, use ChannelRedirect() to connect both existing channels to the bridge, and then join it yourself. This worked for a bit, but the last time I tried it, Asterisk segfaulted. Oops. -- Ian Pilcher arequip...@gmail.com Sometimes there's nothing left to do but crash and burn...or die trying. -- _ -- Bandwidth and Colocation Provided by http://www.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
