Re: [asterisk-users] Polycoms rebooting themselves
On Tuesday 30 August 2011 7:12:54 pm Michael L. Young wrote: > - Original Message - > > > From: "Mike Diehl" > > To: asterisk-users@lists.digium.com > > Sent: Tuesday, August 30, 2011 5:13:22 PM > > Subject: Re: [asterisk-users] Polycoms rebooting themselves > > > > Well, we've taken the time to check out the wiring. It's only 3 > > years old and > > looks like the people who did it knew what they were doing. Nice > > work. > > > > Rebooting the cable modem, router, and switch didn't fix the problem. > > > > Also, we had an instance today where ALL of the phones went down > > within > > minutes of each other. The Internet connection was still active. > > > > Looks like more often than not, all of the phones die at the same > > time. > > > > Any other ideas? > > > > Mike. > > How "latest(ish)" is the firmware? I see in the release notes for 3.3.2 > under corrections: The phones are reporting PolycomSoundPointIP-SPIP_335-UA/3.2.1.0078, so 3.2.1. Looks like upgrading the firmware might be on the menu. However, I'd rather not do this unless there are known issues. > 61147: SoundPoint IP 331, 335, 450, 550, 560, 650, 670: SoundStation IP > 5000: Phone reboots when a GET request is sent to the phone to > /TA/getParam?paramName=reg.1.ringType. > > 68063: Phone reboots when DHCP failover occurs. > > (I know you have 335s but someone else mentioned they had issues with 550s) > 70988: SoundPoint IP 550: Phone when powered by external AC power reboots > during playing of certain audio on full volume. OK, so there may be some known issues I'm still trying to wrap my mind around the fact that they ALL die at once, though. > Just some things that came to mind when I saw your email. I had just > recently reviewed the release notes and they were fresh in my mind. I'll have to do some reading. > Hope you find a fix soon. Thank you. My customer and I are getting pretty frustrated. > Regards, > > Michael > (elguero) Mike Diehl. > -- > _ > -- 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 -- Take care and have fun, Mike Diehl. -- _ -- 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] dahdi with isdn nt_mode, phone no signal still.
On 08/31/2011 04:37 AM, Tamer Higazi wrote: Hi Patrick! I found a NT unit, made the crosscable and connected the NT unit to the ISDN Board which is in NT mode. At the NT unit is the ISDN phone connected. Instead that there is no signal after lifting the phone there is directly a busy tone. Any ideas?! Nope sorry. Check the messages in the asterisk cli or logs. Maybe it tells you what is going on. Regards, 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] dahdi with isdn nt_mode, phone no signal still.
Hi Patrick! I found a NT unit, made the crosscable and connected the NT unit to the ISDN Board which is in NT mode. At the NT unit is the ISDN phone connected. Instead that there is no signal after lifting the phone there is directly a busy tone. Any ideas?! Tamer Am 30.08.2011 20:37, schrieb Tamer Higazi: > Hi Patrick! > Now i got it. > > I am using Gentoo Linux, Asterisk 1.8.5 and Dahdi 2.4.1. > > The patches are automatically integrated at Gentoo. I didn't have to > patch anything. That did the community. > > Another question, I really don't like to buy a new ISDN phone with > external power connector, can I make the power supply for the phone > somehow?! > > Tamer > > Am 30.08.2011 20:05, schrieb Patrick Lists: >> On 08/30/2011 06:32 PM, Tamer Higazi wrote: >>> Hi people! >>> I have managed to set up asterisk 1.8.5. with my 2 ISDN HFC boards on >>> asterisk. On which DAHDI tells me also properly that both of my boards >>> are registered, one in TE and the other on in NT mode. >>> >>> Calls do successfully come inside, but I want to connect my ISDN phone >>> at the board, but the phone is death. What did I make wrong?! >> >> Afaik ISDN phones need power to work. You would need a HFC-S card with a >> power plug where you can hook up power from the computer's power supply >> so the card can power the phone. Like Sangoma's B700. Alternatively >> perhaps you could try to power the phone through an NT1? >> >> Would you mind sharing which version of DAHDI you used and where you got >> the HFC-S patch? >> >> Regards, >> 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 > -- _ -- 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] Good, inexpensive wireless VOIP handsets ?
I found the Uniden UI1868 phones on overstock.com for under $50 each. You'll need to flash them with the latest firmware, but they do the job and they have a wireless handset as well as a base station wired handset. I like that because my family tends to leave the wireless handsets anywhere except the cradle. http://www.overstock.com/Electronics/Uniden-UIP1868-5.8GHz-VoIP-Telephone-System/5594630/product.html On 8/30/2011 9:20 AM, Paul Hayes wrote: On 27/08/11 10:14, Gordon Henderson wrote: On Sat, 27 Aug 2011, Alan Lord (News) wrote: On 26/08/11 19:02, linux guy wrote: I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. We've been using the Siemens Gigaset 685IP range for over three years and I'm (still) very pleased with them: +1 The current generation is the N300 or N300A (A = with answering machine). These have the advantage of being able to do 3 SIP calls at once. You also don't get a handset with it so you can choose whatever handset you want (in theory doesn't even have to be a Gigaset handset as long as it is GAP compatible but you'll have a better time if you do use Gigaset ones, or at least one). These definitely work well with Asterisk for me. So if you really need 6 calls at once then you can get 2 base stations and 6 handsets. If you can live with 6 handsets but only 3 of them in use at once, then a single base will do the job. cheers, Paul. -- _ -- 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 Thanks, -Pablo Sanchez -- You can find me on: Linked In: http://www.linkedin.com/in/sanchezpablo VisualCV: http://www.visualcv.com/users/228823-pabluski/cvs/271611 -- _ -- 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] Polycoms rebooting themselves
- Original Message - > From: "Mike Diehl" > To: asterisk-users@lists.digium.com > Sent: Tuesday, August 30, 2011 5:13:22 PM > Subject: Re: [asterisk-users] Polycoms rebooting themselves > > Well, we've taken the time to check out the wiring. It's only 3 > years old and > looks like the people who did it knew what they were doing. Nice > work. > > Rebooting the cable modem, router, and switch didn't fix the problem. > > Also, we had an instance today where ALL of the phones went down > within > minutes of each other. The Internet connection was still active. > > Looks like more often than not, all of the phones die at the same > time. > > Any other ideas? > > Mike. How "latest(ish)" is the firmware? I see in the release notes for 3.3.2 under corrections: 61147: SoundPoint IP 331, 335, 450, 550, 560, 650, 670: SoundStation IP 5000: Phone reboots when a GET request is sent to the phone to /TA/getParam?paramName=reg.1.ringType. 68063: Phone reboots when DHCP failover occurs. (I know you have 335s but someone else mentioned they had issues with 550s) 70988: SoundPoint IP 550: Phone when powered by external AC power reboots during playing of certain audio on full volume. Just some things that came to mind when I saw your email. I had just recently reviewed the release notes and they were fresh in my mind. Hope you find a fix soon. Regards, Michael (elguero) -- _ -- 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] T.38 passthru on 1.8.5
Txs a lot Kevin. I had just created and account on https://issues.asterisk.org/jira Let me know if this is the right place to post both the pcap capture and the sip logs. If not please help me out creating the account in the right place so that I can provide all the information you need. The sip debug logs I can post here but I need to change the real IPs, which is easy to do because it will be a text file. I appreciate your time and effort in helping us find the roout cause. Fborot From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 13:53:25 -0400 txs a lot for your explanation steve so, it should work w/o spandsp fairly fine if we do not have a bad connection. I see that this version has a lot of fixes related to t.38 but is the implementation already mature enough to guarantee a decent success rate with fax calls? From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 13:15:19 -0400 will installing spandsp help with t.38 pass-through? From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 11:42:41 -0400 both endpoints use public Ips, I just changed the real ones for the privates ones to protect our ips but made a mistake and left the dest as a pub and the orig as private, my bad. but for the record, both are public IPs, there is no nat and iptables is off also, I see that the quintum sends a lot of these packages but asterisk sends only 1 or 2 to the other side. From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 09:44:15 -0400 Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway --> asterisk --> Dialogic IMG 1010 the call starts as a voice call, the remote fax picks up and we hear the fax tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is passed back to the user side [quintum gateway] whcih reply with 200 OK with t.38 and the nothing else happens. After 20 secs of "inactivity" the quintum sends another Invite with voice only and then a BYE. We do see that the quintum sends a lot of messages like this from the quintum's IP [192.168.1.18] but we do not see that asterisk sends the packages to the destination UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) we have this settings on sip.conf faxdetect = yes t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] When we send the fax from the quintum to the Dialogic IMG the fax works 100% of the times. I enabled fax set debug on and udptl set debug on but the console does not show almost anything but the udptl packets shown above. What else should I do?Any ideas/help is greatly appreciated txs a lot fborot -- _ -- 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] Polycoms rebooting themselves
I'm just throwing in my 2c (I don't have polycom) Are your phones auto provisioned then maybe the provisioning server is sending a reboot for some reason or maybe something on the server is sending a sip notify of reboot -Original Message- From: Gord Urquhart Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 30 Aug 2011 15:26:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycoms rebooting themselves -- _ -- 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] Polycoms rebooting themselves
Take a look at the network traffic, things like arp storms etc. A lot of noise on the net can cause reboots. Even if you don't find anything try turning on the storm filter (if it is not on already), its in the Settings -> Advanced-> Administration -> Network settings -> Ethernet I think. g On Tue, Aug 30, 2011 at 2:20 PM, Mike Diehl wrote: > On Tuesday 30 August 2011 3:14:50 pm Tim Nelson wrote: > > - Original Message - > > > > > Well, we've taken the time to check out the wiring. It's only 3 years > > > old and > > > looks like the people who did it knew what they were doing. Nice work. > > > > > > Rebooting the cable modem, router, and switch didn't fix the problem. > > > > > > Also, we had an instance today where ALL of the phones went down > > > within > > > minutes of each other. The Internet connection was still active. > > > > > > Looks like more often than not, all of the phones die at the same > > > time. > > > > > > Any other ideas? > > > > If they're all powered via PoE on the same switch, look to diagnosing the > > switch itself. Look for issues with heat (not enough cooling or > > circulation), or depending on the switch, you could be pulling too much > > power from the PoE module contained within. Does your switch's PoE module > > put out enough power for 'X' number of phones at 'Y' number of watts > each? > > > > Either of these problems would lead to the switch shutting down or > > resetting the PoE module which causes your phone reboots. > > All of the phones are AC powered. Either via an injector or wall outlet; I > don't remember which. Definitely NOT POE. > > > -- > > Take care and have fun, > Mike Diehl. > > -- > _ > -- 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] Transfer to VoiceMail Asterisk 1.6
Hello,
I'm using Asterisk 1.6 with Polycom SoundPoint 650, everything is running
fine except that I can't program a button on Polycom to transfer inbound
call to Voicemail directly.
I have the following in my extension.conf
exten => _547xx,1,Voicemail(${EXTEN:1}@default,u)
Reception can transfer directly to VoiceMail when dialing digit 5 I want to
make a softkey on Polycom 650 does anybody know how to accomplish tranfering
directly to VoiceMail?
Thanks,
Motty
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[asterisk-users] subscriptions from ekiga to asterisk
Hello List!
I have small but strange problem with ekiga 3.2.7 (on Debian Squeeze as
well as on Win XP). For every contact I have listed in ekiga that has a
corresponding hint in the dialplan ekiga tries to subscribe to the hint
but fails. In asterisk the a message like this is shown for every
contact in ekiga:
[Aug 30 23:25:55] NOTICE[30755]: chan_sip.c:21489
handle_request_subscribe: Failed to authenticate device "Michael
Hauptfeld" ;tag=867d4550-bcd1-e011-94fe-f4ec38ab651d
for SUBSCRIBE
I wonder why there is "... 210,hint,SIP/210
exten => 210,1,Dial(SIP/210,${INT_RINGTIME},tTxX)
exten => 210,n,VoiceMail(210,u)
[...]
In sip.conf:
[general]
...
;allowsubscribe=no ; (Default is yes)
;notifyringing = no ; (default: yes)
notifyhold = yes;
notifycid = yes ;
callcounter = yes ;
...
[int-std](!)
type=friend
secret=somesecret
host=dynamic
nat=no
subscribecontext=default
call-limit=10
canreinvite=no
callgroup=8
pickupgroup=8
[210](int-std)
callerid=Hennes <210>
secret=...
[...]
I'm running Asterisk on Debian Squeeze from the debian repositories.
I don't know whether this is a problem of asterisk or ekiga. Seems to be
ekiga, but I hope somebody can help me here.
Sincerly,
Michael
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Re: [asterisk-users] MOH making calls appear hung up
Thanks Danny. I tried that but all that did is make it so when I call the site,
I get hold music instead of ringing. Still has no affect on the call transfer
MOH. :/
Interestingly, the music is playing for about 3-5 seconds before stopping
during the transfer.
I've built all of my phone servers the same at my sites and I'm still sorta
green on some of this stuff. When doing a call transfer, is it using the
macro-stdexten or does it go to the IVR dial plan? Because the entry you noted
is in the IVR dialplan, not the macro-stdexten.
Here's my macro-stdexten:
[macro-stdexten]
exten => s,1,wait(1)
exten => s,2,Dial(${ARG2},20)
exten => s,3,Goto(s-${DIALSTATUS},1)
exten => s,4,Dial(${ARG2},15) ; Ring phone for 15 seconds
exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy
announce
exten => s-BUSY,2,Hangup
exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail
exten => s-NOANSWER,2,Hangup
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${CALLERIDNUM})
exten => a,2,Hangup
Here's my IVR Dialplan:
exten => s,1,Set(TIMEOUT(digit)=4)
exten => s,n,Wait(1)
exten => s,n,Answer
exten => s,n,Dial(SIP/1021,20)
exten => s,n,Wait(1)
exten => s,n,Dial(SIP/6909,15)
exten => s,n,Dial(SIP/6904,15)
exten => s,n,Voicemail(1021,u)
exten => s,n,Hangup
Note: I removed the ",m" because it was only affecting the new incoming calls.
Here's my CLI output:
== Using SIP RTP CoS mark 5
-- Executing [1021@from-PRI:1] Goto("SIP/gw1-0066", "ivr-boi-ntc,s,1")
in new stack
-- Goto (ivr-boi-ntc,s,1)
-- Executing [s@ivr-boi-ntc:1] GotoIfTime("SIP/gw1-0066",
"07:00-17:00,mon-fri,*,*?ivr-boi-ntc-day,s,1") in new stack
-- Goto (ivr-boi-ntc-day,s,1)
-- Executing [s@ivr-boi-ntc-day:1] Set("SIP/gw1-0066",
"TIMEOUT(digit)=4") in new stack
-- Digit timeout set to 4.000
-- Executing [s@ivr-boi-ntc-day:2] Wait("SIP/gw1-0066", "1") in new
stack
-- Executing [s@ivr-boi-ntc-day:3] Answer("SIP/gw1-0066", "") in new
stack
-- Executing [s@ivr-boi-ntc-day:4] Dial("SIP/gw1-0066",
"SIP/1021,20,m") in new stack
== Using SIP RTP CoS mark 5
-- Called 1021
-- Started music on hold, class 'default', on SIP/gw1-0066
-- SIP/1021-0067 is ringing
-- SIP/1021-0067 answered SIP/gw1-0066
-- Stopped music on hold on SIP/gw1-0066
-- Packet2Packet bridging SIP/gw1-0066 and SIP/1021-0067
== Using SIP RTP CoS mark 5
-- Executing [12086597642@from-sip:1] Set("SIP/6908-0068",
"CALLERID(num)=2084331021") in new stack
-- Executing [12086597642@from-sip:2] Dial("SIP/6908-0068",
"SIP/gw1/12086597642,60") in new stack
== Using SIP RTP CoS mark 5
-- Called gw1/12086597642
-- SIP/gw1-0069 is ringing
-- Started music on hold, class 'default', on SIP/gw1-0066
-- SIP/gw1-0069 is making progress passing it to SIP/6908-0068
== Using SIP RTP CoS mark 5
-- Executing [6911@from-sip:1] Macro("SIP/1021-006a",
"stdexten,6911,sip/6911") in new stack
-- Executing [s@macro-stdexten:1] Wait("SIP/1021-006a", "1") in new
stack
-- Executing [s@macro-stdexten:2] Dial("SIP/1021-006a", "sip/6911,20")
in new stack
== Using SIP RTP CoS mark 5
-- Called 6911
-- SIP/6911-006b is ringing
-- Stopped music on hold on SIP/gw1-0066
== Spawn extension (ivr-boi-ntc-day, s, 4) exited non-zero on
'SIP/1021-006a'
== Spawn extension (macro-stdexten, s, 2) exited non-zero on
'SIP/gw1-0066' in macro 'stdexten'
== Spawn extension (from-sip, 6911, 1) exited non-zero on 'SIP/gw1-0066'
-- SIP/gw1-0069 answered SIP/6908-0068
-- Packet2Packet bridging SIP/6908-0068 and SIP/gw1-0069
Thanks,
Kevin Oravits
From: Danny Nicholas [mailto:da...@debsinc.com]
Sent: Tuesday, August 30, 2011 12:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up
Your Dial command stops the MOH - if the command were Dial(SIP/1021,20,m) the
music would continue until connected or timed-out.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 2:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up
I noticed the CLI shows that the music on hold actually stops for some reason?
Here's the output of my CLI:
Connected to Asterisk 1.6.2.19 currently running on localhost (pid = 6363)
Verbosity is at least 28
-- Executing [s@ivr-boi-ntc-day:3] Answer("SIP/gw1-05d6", "") in new
stack
-- Executing [s@ivr-boi-ntc-day:4] Wait("SIP/gw1-05d6", "1") in new
stack
-- Executing [s@ivr-boi-ntc-day:5] Dial("SIP/gw1-05d6", "SIP/1021,20")
in new stack
== Using SIP RTP CoS mark 5
-- Called 1021
-- SIP/1021-05
Re: [asterisk-users] Polycoms rebooting themselves
On Tuesday 30 August 2011 3:14:50 pm Tim Nelson wrote: > - Original Message - > > > Well, we've taken the time to check out the wiring. It's only 3 years > > old and > > looks like the people who did it knew what they were doing. Nice work. > > > > Rebooting the cable modem, router, and switch didn't fix the problem. > > > > Also, we had an instance today where ALL of the phones went down > > within > > minutes of each other. The Internet connection was still active. > > > > Looks like more often than not, all of the phones die at the same > > time. > > > > Any other ideas? > > If they're all powered via PoE on the same switch, look to diagnosing the > switch itself. Look for issues with heat (not enough cooling or > circulation), or depending on the switch, you could be pulling too much > power from the PoE module contained within. Does your switch's PoE module > put out enough power for 'X' number of phones at 'Y' number of watts each? > > Either of these problems would lead to the switch shutting down or > resetting the PoE module which causes your phone reboots. All of the phones are AC powered. Either via an injector or wall outlet; I don't remember which. Definitely NOT POE. -- Take care and have fun, Mike Diehl. -- _ -- 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] Polycoms rebooting themselves
- Original Message - > Well, we've taken the time to check out the wiring. It's only 3 years > old and > looks like the people who did it knew what they were doing. Nice work. > > Rebooting the cable modem, router, and switch didn't fix the problem. > > Also, we had an instance today where ALL of the phones went down > within > minutes of each other. The Internet connection was still active. > > Looks like more often than not, all of the phones die at the same > time. > > Any other ideas? > If they're all powered via PoE on the same switch, look to diagnosing the switch itself. Look for issues with heat (not enough cooling or circulation), or depending on the switch, you could be pulling too much power from the PoE module contained within. Does your switch's PoE module put out enough power for 'X' number of phones at 'Y' number of watts each? Either of these problems would lead to the switch shutting down or resetting the PoE module which causes your phone reboots. --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] Polycoms rebooting themselves
Well, we've taken the time to check out the wiring. It's only 3 years old and looks like the people who did it knew what they were doing. Nice work. Rebooting the cable modem, router, and switch didn't fix the problem. Also, we had an instance today where ALL of the phones went down within minutes of each other. The Internet connection was still active. Looks like more often than not, all of the phones die at the same time. Any other ideas? Mike. On Thursday 18 August 2011 7:58:03 am Justin Sherrill wrote: > I've had mystery reboots with Polycom IP550s - the culprit in both cases > was the network connection. Replacing the cat5 cable to the phone or > changing the attached port fixed it both times. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl > Sent: Wednesday, August 17, 2011 6:01 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Polycoms rebooting themselves > > Hi all, > > I've got a customer with 10 Polycom 335's and the latest(ish) firmware. > For the most part, things are working well. > > However, about once a day, a given phone will just reboot. They don't do > it all at once, and they don't do it along any pattern that I can discern. > > I've got a tcpdump running against one of the phones on my server, but so > far, it's not rebooted, so I've got nothing to look at. > > Any other ideas? -- Take care and have fun, Mike Diehl. -- _ -- 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] USB or Ethernet based FXO device ?
On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez wrote: > Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports. Thanks for the tip. It looks like the smallest option is 8 FXO ports: www.xorcom.com/telephony-interfaces/astribank-models.html -- _ -- 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] MOH making calls appear hung up
Your Dial command stops the MOH - if the command were Dial(SIP/1021,20,m)
the music would continue until connected or timed-out.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 2:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up
I noticed the CLI shows that the music on hold actually stops for some
reason?
Here's the output of my CLI:
Connected to Asterisk 1.6.2.19 currently running on localhost (pid = 6363)
Verbosity is at least 28
-- Executing [s@ivr-boi-ntc-day:3] Answer("SIP/gw1-05d6", "") in new
stack
-- Executing [s@ivr-boi-ntc-day:4] Wait("SIP/gw1-05d6", "1") in new
stack
-- Executing [s@ivr-boi-ntc-day:5] Dial("SIP/gw1-05d6",
"SIP/1021,20") in new stack
== Using SIP RTP CoS mark 5
-- Called 1021
-- SIP/1021-05d7 is ringing
-- SIP/1021-05d7 answered SIP/gw1-05d6
-- Packet2Packet bridging SIP/gw1-05d6 and SIP/1021-05d7
-- Started music on hold, class 'default', on SIP/gw1-05d6
== Using SIP RTP CoS mark 5
-- Executing [6937@from-sip:1] Macro("SIP/1021-05d8",
"stdexten,6937,sip/6937") in new stack
-- Executing [s@macro-stdexten:1] Wait("SIP/1021-05d8", "1") in new
stack
-- Executing [s@macro-stdexten:2] Dial("SIP/1021-05d8",
"sip/6937,20") in new stack
== Using SIP RTP CoS mark 5
-- Called 6937
-- SIP/6937-05d9 is ringing
-- Stopped music on hold on SIP/gw1-05d6
== Spawn extension (ivr-boi-ntc-day, s, 5) exited non-zero on
'SIP/1021-05d8'
-- Nobody picked up in 2 ms
-- Executing [s@macro-stdexten:3] Goto("SIP/gw1-05d6",
"s-NOANSWER,1") in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("SIP/gw1-05d6",
"6937,u") in new stack
-- Playing
'/var/spool/asterisk/voicemail/default/6937/unavail.slin' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'SIP/gw1-05d6' in macro 'stdexten'
== Spawn extension (from-sip, 6937, 1) exited non-zero on
'SIP/gw1-05d6'
Thanks!
Kevin Oravits
From: Danny Nicholas [mailto:da...@debsinc.com]
Sent: Tuesday, August 30, 2011 11:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up
It seems a reasonable likelihood that your moh at the offending site does
not match the codec of the call (IE your MOH is wav and your call codec is
SLIN). Set your verbosity and debug up to 15 and try a call to verify this.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 1:53 PM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] MOH making calls appear hung up
Greetings,
I'm have asterisk servers at about 10 sites, all using Polycom IP 450
phones. With one of my sites, we're having an issue where when a call is
transferred, the MOH is not playing and all the caller is hearing is
silence. The caller of course thinks they have been hung up on, but the call
is actually still in progress and gets successfully transferred if they wait
until the person answers.
I have researched online and even consulted our 3rd party vendor but no one
seems to know how to fix it.
Anyone have any advice? Any help would be appreciated.
Thanks,
Stivaro
--
_
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Re: [asterisk-users] MOH making calls appear hung up
I noticed the CLI shows that the music on hold actually stops for some reason?
Here's the output of my CLI:
Connected to Asterisk 1.6.2.19 currently running on localhost (pid = 6363)
Verbosity is at least 28
-- Executing [s@ivr-boi-ntc-day:3] Answer("SIP/gw1-05d6", "") in new
stack
-- Executing [s@ivr-boi-ntc-day:4] Wait("SIP/gw1-05d6", "1") in new
stack
-- Executing [s@ivr-boi-ntc-day:5] Dial("SIP/gw1-05d6", "SIP/1021,20")
in new stack
== Using SIP RTP CoS mark 5
-- Called 1021
-- SIP/1021-05d7 is ringing
-- SIP/1021-05d7 answered SIP/gw1-05d6
-- Packet2Packet bridging SIP/gw1-05d6 and SIP/1021-05d7
-- Started music on hold, class 'default', on SIP/gw1-05d6
== Using SIP RTP CoS mark 5
-- Executing [6937@from-sip:1] Macro("SIP/1021-05d8",
"stdexten,6937,sip/6937") in new stack
-- Executing [s@macro-stdexten:1] Wait("SIP/1021-05d8", "1") in new
stack
-- Executing [s@macro-stdexten:2] Dial("SIP/1021-05d8", "sip/6937,20")
in new stack
== Using SIP RTP CoS mark 5
-- Called 6937
-- SIP/6937-05d9 is ringing
-- Stopped music on hold on SIP/gw1-05d6
== Spawn extension (ivr-boi-ntc-day, s, 5) exited non-zero on
'SIP/1021-05d8'
-- Nobody picked up in 2 ms
-- Executing [s@macro-stdexten:3] Goto("SIP/gw1-05d6", "s-NOANSWER,1")
in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("SIP/gw1-05d6",
"6937,u") in new stack
-- Playing
'/var/spool/asterisk/voicemail/default/6937/unavail.slin' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'SIP/gw1-05d6' in macro 'stdexten'
== Spawn extension (from-sip, 6937, 1) exited non-zero on 'SIP/gw1-05d6'
Thanks!
Kevin Oravits
From: Danny Nicholas [mailto:da...@debsinc.com]
Sent: Tuesday, August 30, 2011 11:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up
It seems a reasonable likelihood that your moh at the offending site does not
match the codec of the call (IE your MOH is wav and your call codec is SLIN).
Set your verbosity and debug up to 15 and try a call to verify this.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 1:53 PM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] MOH making calls appear hung up
Greetings,
I'm have asterisk servers at about 10 sites, all using Polycom IP 450 phones.
With one of my sites, we're having an issue where when a call is transferred,
the MOH is not playing and all the caller is hearing is silence. The caller of
course thinks they have been hung up on, but the call is actually still in
progress and gets successfully transferred if they wait until the person
answers.
I have researched online and even consulted our 3rd party vendor but no one
seems to know how to fix it.
Anyone have any advice? Any help would be appreciated.
Thanks,
Stivaro
--
_
-- 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:
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Re: [asterisk-users] dahdi with isdn nt_mode, phone no signal still.
On 08/30/2011 08:37 PM, Tamer Higazi wrote: Hi Patrick! Now i got it. I am using Gentoo Linux, Asterisk 1.8.5 and Dahdi 2.4.1. The patches are automatically integrated at Gentoo. I didn't have to patch anything. That did the community. Thanks for the info. Another question, I really don't like to buy a new ISDN phone with external power connector, can I make the power supply for the phone somehow?! You can read more about it here: http://home.foni.net/~jolly1/download/PBX4Linux-2.5.html Scroll all the way down to "2.2 Connect ISDN telephones to your ISDN card" to read howto power your ISDN phone with an NT1 (aka NTBA). You can find a 2nd hand NT1 on ebay or try your local e-market site. I saw several on ebay for $10. Regards, 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] MOH making calls appear hung up
It seems a reasonable likelihood that your moh at the offending site does not match the codec of the call (IE your MOH is wav and your call codec is SLIN). Set your verbosity and debug up to 15 and try a call to verify this. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits Sent: Tuesday, August 30, 2011 1:53 PM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] MOH making calls appear hung up Greetings, I'm have asterisk servers at about 10 sites, all using Polycom IP 450 phones. With one of my sites, we're having an issue where when a call is transferred, the MOH is not playing and all the caller is hearing is silence. The caller of course thinks they have been hung up on, but the call is actually still in progress and gets successfully transferred if they wait until the person answers. I have researched online and even consulted our 3rd party vendor but no one seems to know how to fix it. Anyone have any advice? Any help would be appreciated. Thanks, Stivaro -- _ -- 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] MOH making calls appear hung up
Greetings, I'm have asterisk servers at about 10 sites, all using Polycom IP 450 phones. With one of my sites, we're having an issue where when a call is transferred, the MOH is not playing and all the caller is hearing is silence. The caller of course thinks they have been hung up on, but the call is actually still in progress and gets successfully transferred if they wait until the person answers. I have researched online and even consulted our 3rd party vendor but no one seems to know how to fix it. Anyone have any advice? Any help would be appreciated. Thanks, Stivaro -- _ -- 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] T.38 passthru on 1.8.5
On 08/30/2011 12:53 PM, Fabian Borot wrote: txs a lot for your explanation steve so, it should work w/o spandsp fairly fine if we do not have a bad connection. I see that this version has a lot of fixes related to t.38 but is the implementation already mature enough to guarantee a decent success rate with fax calls? Since you are using T.38, nothing is going to guarantee that... there are many obstacles to achieving it. We can try, though :-) There's really no way to figure out what is happening here without a packet capture of all the traffic and an Asterisk console log at high verbose/debug levels. Is there a way you can produce that and provide it to us without having to reveal confidential information? If not, we can create a private issue on the issue tracker for you to have a place to upload your files without them being visible to the public. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi with isdn nt_mode, phone no signal still.
Hi Patrick! Now i got it. I am using Gentoo Linux, Asterisk 1.8.5 and Dahdi 2.4.1. The patches are automatically integrated at Gentoo. I didn't have to patch anything. That did the community. Another question, I really don't like to buy a new ISDN phone with external power connector, can I make the power supply for the phone somehow?! Tamer Am 30.08.2011 20:05, schrieb Patrick Lists: > On 08/30/2011 06:32 PM, Tamer Higazi wrote: >> Hi people! >> I have managed to set up asterisk 1.8.5. with my 2 ISDN HFC boards on >> asterisk. On which DAHDI tells me also properly that both of my boards >> are registered, one in TE and the other on in NT mode. >> >> Calls do successfully come inside, but I want to connect my ISDN phone >> at the board, but the phone is death. What did I make wrong?! > > Afaik ISDN phones need power to work. You would need a HFC-S card with a > power plug where you can hook up power from the computer's power supply > so the card can power the phone. Like Sangoma's B700. Alternatively > perhaps you could try to power the phone through an NT1? > > Would you mind sharing which version of DAHDI you used and where you got > the HFC-S patch? > > Regards, > 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 -- _ -- 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] dahdi with isdn nt_mode, phone no signal still.
On 08/30/2011 06:32 PM, Tamer Higazi wrote: Hi people! I have managed to set up asterisk 1.8.5. with my 2 ISDN HFC boards on asterisk. On which DAHDI tells me also properly that both of my boards are registered, one in TE and the other on in NT mode. Calls do successfully come inside, but I want to connect my ISDN phone at the board, but the phone is death. What did I make wrong?! Afaik ISDN phones need power to work. You would need a HFC-S card with a power plug where you can hook up power from the computer's power supply so the card can power the phone. Like Sangoma's B700. Alternatively perhaps you could try to power the phone through an NT1? Would you mind sharing which version of DAHDI you used and where you got the HFC-S patch? Regards, 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] T.38 passthru on 1.8.5
txs a lot for your explanation steve so, it should work w/o spandsp fairly fine if we do not have a bad connection. I see that this version has a lot of fixes related to t.38 but is the implementation already mature enough to guarantee a decent success rate with fax calls? From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 13:15:19 -0400 will installing spandsp help with t.38 pass-through? From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 11:42:41 -0400 both endpoints use public Ips, I just changed the real ones for the privates ones to protect our ips but made a mistake and left the dest as a pub and the orig as private, my bad. but for the record, both are public IPs, there is no nat and iptables is off also, I see that the quintum sends a lot of these packages but asterisk sends only 1 or 2 to the other side. From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 09:44:15 -0400 Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway --> asterisk --> Dialogic IMG 1010 the call starts as a voice call, the remote fax picks up and we hear the fax tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is passed back to the user side [quintum gateway] whcih reply with 200 OK with t.38 and the nothing else happens. After 20 secs of "inactivity" the quintum sends another Invite with voice only and then a BYE. We do see that the quintum sends a lot of messages like this from the quintum's IP [192.168.1.18] but we do not see that asterisk sends the packages to the destination UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) we have this settings on sip.conf faxdetect = yes t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] When we send the fax from the quintum to the Dialogic IMG the fax works 100% of the times. I enabled fax set debug on and udptl set debug on but the console does not show almost anything but the udptl packets shown above. What else should I do?Any ideas/help is greatly appreciated txs a lot fborot -- _ -- 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] T.38 passthru on 1.8.5
On 08/31/2011 01:15 AM, Fabian Borot wrote: will installing spandsp help with t.38 pass-through? The only part of spandsp which is relevant to T.38 passthrough is its modem tone detection module, and I don't think the standard Asterisk distribution can make use of that. Some people do use it, to overcome the limitations in Asterisk's own tone detection, but I don't think they make their patches available. Steve From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 11:42:41 -0400 both endpoints use public Ips, I just changed the real ones for the privates ones to protect our ips but made a mistake and left the dest as a pub and the orig as private, my bad. but for the record, both are public IPs, there is no nat and iptables is off also, I see that the quintum sends a lot of these packages but asterisk sends only 1 or 2 to the other side. From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 09:44:15 -0400 Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway --> asterisk --> Dialogic IMG 1010 the call starts as a voice call, the remote fax picks up and we hear the fax tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is passed back to the user side [quintum gateway] whcih reply with 200 OK with t.38 and the nothing else happens. After 20 secs of "inactivity" the quintum sends another Invite with voice only and then a BYE. We do see that the quintum sends a lot of messages like this from the quintum's IP [192.168.1.18] but we do not see that asterisk sends the packages to the destination UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) we have this settings on sip.conf faxdetect = yes t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] When we send the fax from the quintum to the Dialogic IMG the fax works 100% of the times. I enabled fax set debug on and udptl set debug on but the console does not show almost anything but the udptl packets shown above. What else should I do?Any ideas/help is greatly appreciated txs a lot fborot -- _ -- 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] T.38 passthru on 1.8.5
will installing spandsp help with t.38 pass-through? From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 11:42:41 -0400 both endpoints use public Ips, I just changed the real ones for the privates ones to protect our ips but made a mistake and left the dest as a pub and the orig as private, my bad. but for the record, both are public IPs, there is no nat and iptables is off also, I see that the quintum sends a lot of these packages but asterisk sends only 1 or 2 to the other side. From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 09:44:15 -0400 Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway --> asterisk --> Dialogic IMG 1010 the call starts as a voice call, the remote fax picks up and we hear the fax tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is passed back to the user side [quintum gateway] whcih reply with 200 OK with t.38 and the nothing else happens. After 20 secs of "inactivity" the quintum sends another Invite with voice only and then a BYE. We do see that the quintum sends a lot of messages like this from the quintum's IP [192.168.1.18] but we do not see that asterisk sends the packages to the destination UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) we have this settings on sip.conf faxdetect = yes t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] When we send the fax from the quintum to the Dialogic IMG the fax works 100% of the times. I enabled fax set debug on and udptl set debug on but the console does not show almost anything but the udptl packets shown above. What else should I do?Any ideas/help is greatly appreciated txs a lot fborot -- _ -- 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 with isdn nt_mode, phone no signal still.
Hi people! I have managed to set up asterisk 1.8.5. with my 2 ISDN HFC boards on asterisk. On which DAHDI tells me also properly that both of my boards are registered, one in TE and the other on in NT mode. Calls do successfully come inside, but I want to connect my ISDN phone at the board, but the phone is death. What did I make wrong?! dahdi_scan: office tamer # dahdi_scan [1] active=yes alarms=OK description=HFC-S PCI A ISDN card 0 [TE] name=ZTHFC1 manufacturer=Cologne Chips devicetype=HFC-S PCI-A ISDN location=PCI Bus 01 Slot 01 basechan=1 totchans=3 irq=17 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=AMI framing_opts=CCS coding=AMI framing=CCS [2] active=yes alarms=OK description=HFC-S PCI A ISDN card 1 [NT] name=ZTHFC2 manufacturer=Cologne Chips devicetype=HFC-S PCI-A ISDN location=PCI Bus 01 Slot 02 basechan=4 totchans=3 irq=21 type=digital-NT syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=AMI framing_opts=CCS coding= framing=CAS office tamer # and dahd status in asterisk: office*CLI> dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO HFC-S PCI A ISDN card 0 [TE] OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) HFC-S PCI A ISDN card 1 [NT] OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) TE-mode works fine so far. -- _ -- 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] T.38 passthru on 1.8.5
both endpoints use public Ips, I just changed the real ones for the privates ones to protect our ips but made a mistake and left the dest as a pub and the orig as private, my bad. but for the record, both are public IPs, there is no nat and iptables is off also, I see that the quintum sends a lot of these packages but asterisk sends only 1 or 2 to the other side. From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 09:44:15 -0400 Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway --> asterisk --> Dialogic IMG 1010 the call starts as a voice call, the remote fax picks up and we hear the fax tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is passed back to the user side [quintum gateway] whcih reply with 200 OK with t.38 and the nothing else happens. After 20 secs of "inactivity" the quintum sends another Invite with voice only and then a BYE. We do see that the quintum sends a lot of messages like this from the quintum's IP [192.168.1.18] but we do not see that asterisk sends the packages to the destination UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) we have this settings on sip.conf faxdetect = yes t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] When we send the fax from the quintum to the Dialogic IMG the fax works 100% of the times. I enabled fax set debug on and udptl set debug on but the console does not show almost anything but the udptl packets shown above. What else should I do?Any ideas/help is greatly appreciated txs a lot fborot -- _ -- 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] USB or Ethernet based FXO device ?
On Tue, 2011-08-30 at 01:31 +0200, Gilles wrote: > On Sat, 27 Aug 2011 09:31:12 -0600, linux guy > wrote: > >I'm looking for an FXO device to connect to a POTS line that communicates > >via USB or Ethernet. > > For USB, AFAIK, there's only the one from Sangoma. All others are > Ethernet-based. > > www.voip-info.org/wiki/view/VoIP+Gateways > Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] T.38 passthru on 1.8.5
Hi, Am Dienstag, den 30.08.2011, 09:44 -0400 schrieb Fabian Borot: > > Hello > We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk > 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running > Linux on 2011-08-26 21:31:22 UTC] > > The call flow is: > quintum gateway --> asterisk --> Dialogic IMG 1010 > > the call starts as a voice call, the remote fax picks up and we hear > the fax tone, the we see the re-invite from the IMG asking for t.38, > the RE-Invite is passed back to the user side [quintum gateway] whcih > reply with 200 OK with t.38 and the nothing else happens. After 20 > secs of "inactivity" the quintum sends another Invite with voice only > and then a BYE. > > We do see that the quintum sends a lot of messages like this from the > quintum's IP [192.168.1.18] but we do not see that asterisk sends the > packages to the destination > > UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, > seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) > > we have this settings on sip.conf > faxdetect = yes > t38pt_udptl = yes,maxdatagram=400 [I have tested with several > combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] > > When we send the fax from the quintum to the Dialogic IMG the fax > works 100% of the times. > I enabled fax set debug on and udptl set debug on but the console does > not show almost anything but the udptl packets shown above. > What else should I do?Any ideas/help is greatly appreciated I assume there is some NAT/firewall on the way? (The 192.30.189.146 is public internet and 192.168.1.18 is RFC1918). Be sure to let udptl pass through your firewall/NAT. Udptl is using a port range of its own (different from the range used for audio). Audio is always symmetric. Audio in one directions implies audio in the opposite direction. A NAT Gateway open for one direction is normally no problem for audio. T.38 is not symmetric. One side sends and waits for some kind of ACK. The other side waits for data. If the sending side gets through the NAT Gateway all is fine. If the sending side is not getting through, the transmission is aborted (after timeout). HTH, Karsten -- _ -- 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] Avaya to Asterisk Voice mail
Search the forum - I believe I remember a recent exchange on this subject From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dustin fails Sent: Tuesday, August 30, 2011 10:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Avaya to Asterisk Voice mail Has anyone have Avaya setup to ring to Asterisk voice mail over an analogue line. The issue I am having is Avaya is sending the originating caller id not the station id so Asterisk see that originating id so I can't route the call correctly in Asterisk. Thanks! Dustin -- _ -- 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] Avaya to Asterisk Voice mail
Has anyone have Avaya setup to ring to Asterisk voice mail over an analogue line. The issue I am having is Avaya is sending the originating caller id not the station id so Asterisk see that originating id so I can't route the call correctly in Asterisk. Thanks! Dustin -- _ -- 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] same sip peer as user and provider
yes, same thing From: fbo...@hotmail.com To: fbo...@hotmail.com Subject: RE: same sip peer as user and provider Date: Tue, 30 Aug 2011 10:35:01 -0400 yes my friend. same thing From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: same sip peer as user and provider Date: Tue, 30 Aug 2011 10:16:11 -0400 Hello Up to version 1.6.0 we have been able to configure the same SIP device as a user [inbound trunk] and as a peer [outbound trunk] w/o issues. After we switched to version 1.8 this setup wont work, apparently one can not have the same IP on 2 different trunks anymore. The trunk that is configured as user or friend is not choosen when the inbound call hits asterisk, instead the outbound trunk is and that trunk is usually w/o context and then asterisk can not find any "call logic" in the dialplan in the default extension, hence the call fails. as a workaround I have been trying the SIP_CODEC variables [inbound and outbound] but it wont help me in all cases. Also I can not set the ptime on the fly using those variables in the dialplan. After reading in the forums and the books/guides apparently the "users" are matched by the From header and "peers" are matched by IP. Is this is the intended behavior now? any help is greatly appreciated, txs a lot in advance fborot -- _ -- 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] same sip peer as user and provider
On Tuesday 30 August 2011, Fabian Borot wrote: > Up to version 1.6.0 we have been able to configure the same SIP device as > a user [inbound trunk] and as a peer [outbound trunk] w/o issues. After we > switched to version 1.8 this setup wont work, apparently one can not have > the same IP on 2 different trunks anymore. Have you tried defining the device as "type=friend" ? -- AJS Answers come *after* questions. -- _ -- 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] same sip peer as user and provider
Hello Up to version 1.6.0 we have been able to configure the same SIP device as a user [inbound trunk] and as a peer [outbound trunk] w/o issues. After we switched to version 1.8 this setup wont work, apparently one can not have the same IP on 2 different trunks anymore. The trunk that is configured as user or friend is not choosen when the inbound call hits asterisk, instead the outbound trunk is and that trunk is usually w/o context and then asterisk can not find any "call logic" in the dialplan in the default extension, hence the call fails. as a workaround I have been trying the SIP_CODEC variables [inbound and outbound] but it wont help me in all cases. Also I can not set the ptime on the fly using those variables in the dialplan. After reading in the forums and the books/guides apparently the "users" are matched by the From header and "peers" are matched by IP. Is this is the intended behavior now? any help is greatly appreciated, txs a lot in advance fborot -- _ -- 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] T.38 passthru on 1.8.5
Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway --> asterisk --> Dialogic IMG 1010 the call starts as a voice call, the remote fax picks up and we hear the fax tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is passed back to the user side [quintum gateway] whcih reply with 200 OK with t.38 and the nothing else happens. After 20 secs of "inactivity" the quintum sends another Invite with voice only and then a BYE. We do see that the quintum sends a lot of messages like this from the quintum's IP [192.168.1.18] but we do not see that asterisk sends the packages to the destination UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) we have this settings on sip.conf faxdetect = yes t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] When we send the fax from the quintum to the Dialogic IMG the fax works 100% of the times. I enabled fax set debug on and udptl set debug on but the console does not show almost anything but the udptl packets shown above. What else should I do?Any ideas/help is greatly appreciated txs a lot fborot -- _ -- 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] Good, inexpensive wireless VOIP handsets ?
On 27/08/11 10:14, Gordon Henderson wrote: On Sat, 27 Aug 2011, Alan Lord (News) wrote: On 26/08/11 19:02, linux guy wrote: I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. We've been using the Siemens Gigaset 685IP range for over three years and I'm (still) very pleased with them: +1 The current generation is the N300 or N300A (A = with answering machine). These have the advantage of being able to do 3 SIP calls at once. You also don't get a handset with it so you can choose whatever handset you want (in theory doesn't even have to be a Gigaset handset as long as it is GAP compatible but you'll have a better time if you do use Gigaset ones, or at least one). These definitely work well with Asterisk for me. So if you really need 6 calls at once then you can get 2 base stations and 6 handsets. If you can live with 6 handsets but only 3 of them in use at once, then a single base will do the job. cheers, Paul. -- _ -- 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] FREE webinar video about Auto-Dialer Business Model (Telemarketing)
On 11-08-30 05:16 AM, Kolmisoft Marketing wrote: NOTE: This is not attempt to sell you anything. No product or service is sold/marketed in the video. That maybe the case, but this is still a non-commercial mailing list. Please use asterisk-biz in the future. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted a modified SIP message body
On 08/30/2011 07:36 AM, Jaime Lozano wrote: I have been using wireshark to capture some traffic. I'm talking when the PBX sends OK (200) connection accepted. 3Com PBX sends "TZ=7200\n" (an much more things) in a SIP packet message body but Asterisk PBX sends packets without message body, it only sends variables in the message header. So I want Asterisk to send packets with a message body and its proper content. This is extremely confusing, to say the least. 'a message body' and 'its proper content' are ambiguous, especially since Asterisk already works with pretty much every SIP UA on the planet and none of them require the things you are asking for. Why don't you post an actual example of what you think Asterisk should be sending, and what it is actually sending, rather than trying to describe the differences (which is clearly not going well)? In general, though, you can't just put random content in a SIP request or response message body; the message body is usually of a defined type (application/sdp, for example), and has rules about what it can and cannot contain. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted a modified SIP message body
Hello, I have been using wireshark to capture some traffic. I'm talking when the PBX sends OK (200) connection accepted. 3Com PBX sends "TZ=7200\n" (an much more things) in a SIP packet message body but Asterisk PBX sends packets without message body, it only sends variables in the message header. So I want Asterisk to send packets with a message body and its proper content. I've been looking for, sure spending my time. Modifying the source code is not very realistic, but i have to try. The file channels/chan_sip.c seems to have interesting functions like add_header(), add_tcodec_to_sdp(), handle_response_invite(), handle_response(), also struct cfalias. I know C programming but it's really hard to understand the code. Should I ask in the developers list? have a nice day 2011/8/27 Jaime Lozano > > > -- Forwarded message -- > From: Olle E. Johansson > Date: 2011/8/26 > Subject: Re: [asterisk-users] Wanted a modified SIP message body > To: Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > > > > 26 aug 2011 kl. 14:06 skrev Jaime Lozano: > > > Hello, > > In which file do I use SIPAddHeader()? > > Please consider that the packet goes from the PBX to the telephone, and > what I want is not a header because the "TZ: 7200\n" is in the *message > body* not in the *message header*. > > That's no longer a SIP header, it's part of the SDP you want to change. You > can't do that without changing the source code. > > /O > -- > _ > -- 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] Possible Bug? .call files executing multiple times
Thanks Danny. Changing the ownership of the .call files seems to have fixed the problem and I can now see that asterisk is adding a "StartRetry" line to the end of the file after it makes the first call, which it was unable to do before since the file was owned by root: cp test5703.call /tmp/test.call && chown asterisk:asterisk /tmp/test.call && mv /tmp/test.call /var/spool/asterisk/outgoing/ Thanks, Brandon On 08/29/2011 05:41 PM, Danny Nicholas wrote: Asterisk has to be able to execute and rewrite the file - the call file is updated in place and when the call is considered successful, removed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 4:28 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times Also I should note that we use the 'noatime' attribute on the /var filesystem, would this cause the problem below? On 08/29/2011 05:22 PM, Brandon Phelps wrote: Here is the contents of the .call file. The file is the same before the move as after (I did a cat on the file after the move, while the phone was ringing a second time): Channel: Local/5703@ext-main Callerid: "MyCompany"<8005551234> Set: TicketNumber=100 Set: CallerID_Num=8005551234 Set: CALLSTATUS=0 Context: ext-autodialer MaxRetries: 0 WaitTime: 45 Extension: s Priority: 1 We have tried using a SIP channel as well (as opposed to Local) with the same results. The s extension of ext-autodialer runs an AGI script which makes use of those Set: variables. I can most easily reproduce the problem by simply not answering the call. After 2 or 3 rings line 2 on the phone lights up indicating another call. If I reject the first call and answer the second call, it's the same script. Also during my most recent test the following happened: 1. I moved file to /var/spool/asterisk/outgoing 2. Phone rang on line 1 3. I let phone continue to ring 4. After 3 rings, line 2 started ringing (another call from the same .call file) 5. I rejected both calls, sending both to voicemail. 6. 6 or 7 seconds after rejecting both calls, the phone rang a 3rd time. 7. I let the phone ring until it was automatically moved to voicemail and finally the .call file was removed. On 08/29/2011 11:00 AM, Danny Nicholas wrote: Can you post the .call file (with called number blacked out) before call and after 1-2 calls? (file 1 should be before you mv to /v/s/a/o, file 2 should be from /v/s/a/o). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 8:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times On 08/19/2011 09:14 AM, Brandon Phelps wrote: Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call&& mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Sorry to bring this back up but I am still having this issue and haven't had any luck resolving it. It should be noted that the .call files in question are set to MaxRetries: 0, and simply connect the call to the 's' extension in a custom context. From there the context is pretty complicated, running some AGI scripts along with some dealing with user input, basically a simple IVR. Any help would be appreciated. Thanks, Brandon -- _ -- 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 Asteris
[asterisk-users] FREE webinar video about Auto-Dialer Business Model (Telemarketing)
Hello, We would like to share our webinar about Auto-Dialer Business Model (Telemarketing). It is educational video which we made for our clients and now we are sharing it with you. http://www.kolmisoft.com/how-to-start-a-VoIP-business/webinars/ Enjoy. NOTE: This is not attempt to sell you anything. No product or service is sold/marketed in the video. Regards, Kolmisoft Team www.kolmisoft.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] seeding an originated number in a SIP phone [was: Re: Thunderbird extension using AMI to dial]
29 aug 2011 kl. 15:05 skrev Kevin P. Fleming: > On 08/28/2011 01:56 AM, Tzafrir Cohen wrote: >> On Thu, Aug 25, 2011 at 07:36:53PM +0100, Chris Hastie wrote: >>> Hi >>> >>> I've just added direct support for AMI to a forthcoming version of >>> TBDialOut, a Thunderbird extension for dialling direct from >>> Thunderbird's address book. If anyone fancies testing it I'd be grateful >>> for any feedback. If you feel like casting a critical eye over the code, >>> or doing some translating, even better. >>> >>> AMI support is available in TBDialOut 1.7.0pre1, which can be found >>> either at http://www.oak-wood.co.uk/tbdialout/ or from the 'Development >>> channel' at the bottom of the page at >>> https://addons.mozilla.org/en-US/thunderbird/addon/tbdialout/ >> >> We already have a dialer script (sent to this list a while ago) so it's >> good to see that this extension support that simpler option as well (I >> don't use ThunderBird, as you can see. Some others in the office do use >> it). >> >> One followup question: I originate a call from a SIP phone to some >> remote number. The problem is that the number will not show up properly >> in the list of outgoing calls for the phone. Any idea how to fix this >> (for whatever SIP phone)? > > You aren't originating a call *from* the phone (that would require some sort > of API into the phone itself to make it place a call). You are originating a > call *to* the phone and also to another endpoint; as far as the SIP phone is > concerned, this is an incoming call. > > I've never seen discussion of a desire to provide a method for an incoming > call to be treated as if the endpoint had placed the call itself in any of > the SIP discussion lists I frequent... so I'm pretty sure there's no standard > way to do this. Oh, there is - REFER. We could possibly implement sending a REFER request to the phone, something that is frequently used to do call setups from click-to-call apps. This is not something we do support in Asterisk today. I've implemented it using SIP libraries since Asterisk doesn't have to be involved in the REFER. If you do ORIGINATE from the phone you have to be aware that Asterisk lacks some security framework here. An application that has ORIGINATE access can reach the whole dialplan. I have patches for that which needs to be moved forward. My proposal is to add a default context to manager accounts to put a limitation of destinations they can reach with originate and redirect AMI commands (which where the only ones I could come up with as dangerous in this regard). /O -- _ -- 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
