[asterisk-users] Ringing detection ?
Hi !
I have an application where I originate a call with a call file and play
some pre-recorded message when the person answers. And it's working
correctly.
Now, I've been asked to add the support for extenstion numbers.
I've been able to actualy send the extension numbver via the SendDTMF
command. It works perfectly. But after the dtmf have been sent, the
dialplan shoud wait for someone to answer.
The problem is that during that time, the phone is rining. So withing
north-america, it's typicly ring 2 secs, silence 4 seconds.
So, I could use AMD() WaitForSilence(4100) for exemple. But that would
require the person at the other end to be silent for 4 seconds. That's
unrealistic.
So, I'm searching for a way for my dialplan to detect ringing and only
lauch Amd/WaitforSilence after the person answers...
My curent ael dialplan for my originated call is :
500 = {
Answer();
Wait (1);
if (${LEN(${noPoste})})
{
SendDTMF(${noPoste});
}
Background(silence/1);
AMD();
WaitForSilence(500);
for (x=0; ${x} 3; x=${x} + 1) {
Background(outcall/outcall-${idJob});
Background(outcall-confirm);
WaitExten(5);
};
goto diffuseurappel|3|1;
};
Any ideas ?
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[asterisk-users] Problem with libpri / asterisk
Hi all !
We currently have an asterisk box that is rather old (runs Asterisk
1.4.21.2), and it's connected to the PSTN with a sangoma A104d card.
Now we have a new PRI at another location, and I use that occasion to
build 2 new servers, one to replace our aging one and a new one for this new
pri.
So I downloaded the lastest libpri / asterisk / wanpipe driver, but the
previous version of dahdi (2.5), since the latest wanpipe isn't compatible
with dahdi 2.6. All is built from source
Now, all seems to be working OK. I can connect a SIP phone to my new box,
make calls to the outside, receive calls etc.
But, I can't seem to bridge a call. So on my new server, with the new PRI, I
got a Sangoma a104 card (no echo-canceler on this one).
In my extensions.ael, I got this :
418nx1 = {
Answer();
Wait (2);
Playback(demo-thanks);
Dial(${TRUNK}/418nx2);
};
TRUNK is DAHDI/G1
Where 418nx1 is a DID on my new PRI and 418nx2 is my cellphone
number.
When I do a call from my home phone or cell phone to my new PRI to
418nx1, I hear the demo-thanks file, and then it dials out. My cellphone
rings, but as soon as I pick up the call, the calls hangs up :
-- Accepting call from '418nx2' to '418nx1' on channel 0/1, span
1
-- Executing [418nx1@ael-default:1] Answer(DAHDI/i1/418nx2-b,
) in new stack
-- Executing [418nx1@ael-default:2] Wait(DAHDI/i1/418nx2-b,
2) in new stack
-- Executing [418nx1@ael-default:3]
Playback(DAHDI/i1/418nx2-b, demo-thanks) in new stack
-- DAHDI/i1/418nx2-b Playing 'demo-thanks.ulaw' (language 'fr')
-- Executing [418nx1@ael-default:4] Dial(DAHDI/i1/418nx2-b,
DAHDI/G1/418nx2) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called DAHDI/G1/418nx2
-- DAHDI/i1/418nx2-c is proceeding passing it to
DAHDI/i1/418nx2-b
-- DAHDI/i1/418nx2-c is ringing
-- DAHDI/i1/418nx2-c is making progress passing it to
DAHDI/i1/418nx2-b
-- DAHDI/i1/418nx2-c answered DAHDI/i1/418nx2-b
-- Native bridging DAHDI/i1/418nx2-b and DAHDI/i1/418nx2-c
-- Span 1: Channel 0/1 got hangup request, cause 16
-- Hungup 'DAHDI/i1/418nx2-c'
== Spawn extension (ael-default, 418nx1, 4) exited non-zero on
'DAHDI/i1/418nx2-b'
-- Hungup 'DAHDI/i1/418nx2-b'
BUT, if I originate the call from my curent PRI, it goes in and out and all
is well. I noticed that if the calls go trough correctly and hangup
manually, it also stats the exact same thing (cause 16). So the above
console output might not be that much usefull...
I've had a case open with Sangoma for this issue, and they suggested I go
the libpri/asterisk for more help debuging this issue, since on their end,
the disconnect comes from the telco...
They suggested I try a different version of asterisk, wich I did to no
avail, or try there NBE product instead of libpri...
So, did anybody ever encontered something like that ? What steps should I
take to diagnose the problem furhter ?
Thanks for any help.
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Re: [asterisk-users] Problem with libpri / asterisk
(...) My guess is your new setup is trying to do a PRI 2B Transfer (meaning that Asterisk is trying to handoff two B channels of a PRI to the upstream switch). It is probably being rejected and the call is hanging up. You will need to dig into the PRI debug of both scenarios and compare. I was not even aware that Asterisk could do that so it may be some new feature being worked on. I just found this: http://wiki.sangoma.com/Asterisk-FAQ#TBCT Maybe you should check and see if it is enabled. My god ! That was it ! It was enabled, I had transfer = yes, but there was no mention of facilityenable. I disabled it, restarted asterisk, and voilà ! Thanks for pointing that out ! -- _ -- 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] Trying out a new version with sangoma card
In asterisk CLI do pri show spans. The fact the card is in RED alert means the hardware does not see the pri line connected to the card. I probably made a mistake in copying / pasting. pri show spans was showing something like : PRI span 1/0: Provisioned, Up, Active Calls can enter, I see them arriving on the console, but they imediatly got hangun, cause 6. -- _ -- 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] Trying out a new version with sangoma card
Show us the output of a failed call with pri debug enabled on that span. It will be difficult, since the PRI is in use on our old asterisk box. I will have to get to the colo at night, to avoid disrupting calls during the day. Is there any other thing that I should collect ? -- _ -- 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] Trying out a new version with sangoma card
Le 2011-05-09 09:31, Jim Dickenson a écrit : Make sure the firmware on the card is latest. I had a problem, not like your, and flashing the card to the latest firmware resolved it. It appears it did not change anything... So, to re-cap, I have a sangoma A101 card, with the firmware uptodate, on the asterisk 1.8.3.3, dahdi-linux 2.4.1.2, and dahdi-tools 2.4.1. When asterisk is running, cat /proc/dahdi/1 yields : Span 1: WPT1/0 wanpipe1 card 0 (MASTER) B8ZS/ESF RED 1 WPT1/0/1 Clear (In use) 2 WPT1/0/2 Clear (In use) (...) 24 WPT1/0/24 Hardware-assisted HDLC (In use) And when it's not, the (In use) go away. When, dialing I get Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) So, does anybody got any idea ? -- _ -- 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] Trying out a new version with sangoma card
Hi ! We curently have a centos 5 / asterisk 1.4 server that we have some DTMF problems with. It has a Sangoma A104d card and only port one is used to connect to the PSTN. Port 2 is conencted via a cross-over cable to a RAS for modem access and port 3 is connected for data communication via PPP. Now, I want to freshen this setup to something newer. So I installed a Scientific Linux 6 server, with asterisk 1.8 and the latest Sangoma drivers and an A101 card I had laying around. I did a test this weekend and pluged in our PRI in that test server. I never got succeded to have a call trough. When I dialed in, the call is hanged up with : Channel 1/1, span 1 got hanup, cause 6 Spawn extension (ael-default, s, 3) exited non-zero on 'DAHDI/i1/NPANXX-2' Hungup 'DAHDI/i1/NPANXX-2' Here's my dahdi/system.conf : loadzone=us defaultzone=us #Sangoma A101 port 1 [slot:0 bus:6 span:1] wanpipe1 span=1,1,0,esf,b8zs bchan=1-23 echocanceller=mg2,1-23 hardhdlc=24 my asterisk/chan_dahdi.conf is rather long, but is mostly defaults, with : switchtype=national pridialplan=unknown signalling=pri_cpe group=1 channel = 1-23 as the last non-commented lines. So, for one thing, the card I have in my test server doesn't have an hardware echo canceller, but it's still enabled in my wanpip setting. Could that be a source of problem ? Other than that, is there anything obvious I've missed ? -- _ -- 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] Trying out a new version with sangoma card
Make sure the firmware on the card is latest. I had a problem, not like your, and flashing the card to the latest firmware resolved it. I did the upgrade, I will make another test when appropriate. I will also upgrade my curent card, I am curent at version 25, wich dates 2007, it might solve our curent problem also... 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] Under heavy attack
I just wanted to add my voice to this attack. I saw the morning that I had 200+ distinct ips since the weekend. I used a small perl script that blocks failed usernames and passwords at iptables level I found thei morning : http://www.teamforrest.com/blog/171/asterisk-no-matching-peer-found-block/ Regards, My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- _ -- 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] Pickup parcked call from Aastra 9480i ct cordless
Here, we have 2 aastra 9133i and on 9480i ct cordless. We use often the park call feature of asterisk to transfer calls to one another. But the 9480i ct cordless cannot pickup a parked call. When manually entering 701 (parked call extention), the phone display Call failed (appel écoué in french). Nothing is displayed on the asterisk console. When doing it from other aastra phones (same config), or other make phones, it works. Any hints on possible causes ? -- _ -- 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] Pickup parcked call from Aastra 9480i ctcordless
Make sure your verbosity is set to at least 5 and try to see the CLI output on failure again. Are you sure the call is parked on 701 (not 702-720 as defined in features.conf)? Yes, after I can pick it up from my phone (9133i), and it works. I had verbosity at 6 at the moment of testing. When he enters 701, only his phones displays Failed, nothing in asterisk. I can pickup after that on mine. Other calls/extension dials work from this phone? Yes, our extensions are 3 digits, and all other calls to / from this (problematic) phone works. Regads, -- _ -- 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] Pickup parcked call from Aastra9480i ctcordless
Is the phone defined as a SIP extension/peer? If so, try sip set debug peer xxx and try the call/pickup again. Yes, and doing so, the phone could no longer dial out, bizare. Yes, after I can pick it up from my phone (9133i), and it works. I had verbosity at 6 at the moment of testing. When he enters 701, only his phones displays Failed, nothing in asterisk. I can pickup after that on mine. Is there a dial plan on the phone that you need to alter? Yes : x+#|xx+*, same as mine. After the problem with dial-out stated on top, we rebooted the phone another time, and now everithing works... That was strange... -- _ -- 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] Redone setup, bizare problems
Hi ! Sorry if this is a long post... I had this setup for about a year without problems : Network A - wrv200 - internet - wrv200 - net b The 2 networks are linked with an ipsec vpn. The 2 internet connections are with the same cable company to minimize latency, both separates /24 subnets. On network A, I got 2 computers, a single sip phone (aastra 9112i). On network B, I got a linux box acting as smb domain controler and many other thins, another linux box that only do an asterisk server, 4 computers, 3 sip phones (2 aastra 9112i and one 480i ct), 1 analog ATA for the alarm system, an internet point of sale terminal, that's about it. The asterisk server is connected to the PSTN with a VOIP provider trough IAX. All of this was working fine. Since the last couple of months, we had some qos problems with the exterior. The sound was choping up randomly. With what I can do with the linksys wrv200, I was limited. I also had some filesystem issues that made the asterisk server locks up from time to time. So I wanted to reformat the server and re-install it on a ssd drive. So for the time of the re-format, I installed asterisk on my smb server, and re-copied /etc/asterisk, /var/spool/asterisk and /var/lib/asterisk to it. When I arrived at home, the externel power supply of the little shuttle box refused to power up. So I ended up building a new computer from scratch. This new server will be acting as asterisk server and router, replacing the wrv200 from network b (demoting it to wireless access point and switch). Last sunday, I installed the server, re-copying the 3 folders of asterisk in the same maner. I then had a hard time making the phones register to the server. I always had no service with the mwi light steady on. I finaly got the phones to register, I'm still not sure exactly what I did to make it work. And the phone on network A didn't work wither. I'll get to that one later. The linksys pap2t ata didn't had the problem, it registered right as I started the server. On the phones on network B, since then I get the MWI light come on momentarely, with the no service on the display, and then all comes back to normal. But all phones can make and receive calls. For the remote phone, for those familiar with ipsec vpns, it's a net to net connection. So, the gateway cannot reach the computers on the network on the other side. So I had to add a static route on my router/asterisk server to be able to reach the phone on the other side. It was able to register for some time, but the next morning it was no service, and I wasn't able to make it work again. I ended up connecting the phone trough the externel interface of the router/asterisk server. So, my questions : Why is the phones are constatly showing no service with the light flashing ? Is there a way for the remote phone from net A to connect properly trough the vpn ? Thanks for any hints, Niolas -- _ -- 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] ATA for large networks
I personnlay found that marc is better than google when searching mailing lists : http://marc.info/?l=asterisk-usersr=1w=2 What is the best-recommended resource for searching archives of this mailing list? Thanks for your time ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer a call without announce : no sound
When we receive a call from outside (via a sangoma 104d card) and we do a blind transfer, that is without anouncing to the called party , we have no sound either way. Exemple : I take my cell phone to call my * box, it rings on my aastra 9113i phone, I answer. Then I hit the xfer buton, make my second call to another extention (it can be either a aastra phone, nortel phone trough ciel portico, whatever. As soon it rings I hangup or hit the xfer buton again. Then the bridged call between the other extension and the zap channel have no sound either way. If I wait for the called party to answer and announce the transfer, all is fine. I've had report of sound one way also, but I wasn't able to reproduce. Here's the log from my console : -- SIP/224-09e0f098 answered Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Executing [EMAIL PROTECTED]:1] Macro(SIP/224-09e1d728, ael-std-exten|225|SIP/225) in new stack -- Executing [EMAIL PROTECTED]:1] Set(SIP/224-09e1d728, ext=225) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/224-09e1d728, dev=SIP/225) in new stack -- Executing [EMAIL PROTECTED]:3] Answer(SIP/224-09e1d728, ) in new stack -- Executing [EMAIL PROTECTED]:4] NoOp(SIP/224-09e1d728, Nicolas Ross 224) in new stack -- Executing [EMAIL PROTECTED]:5] Wait(SIP/224-09e1d728, 0.5) in new stack -- Executing [EMAIL PROTECTED]:6] Dial(SIP/224-09e1d728, SIP/225|15) in new stack -- Called 225 -- SIP/225-09e73388 is ringing -- Stopped music on hold on Zap/1-1 == Spawn extension (macro-ael-std-exten, s, 6) exited non-zero on 'SIP/224-09e1d728ZOMBIE' in macro 'ael-std-exten' == Spawn extension (macro-ael-std-exten, s, 6) exited non-zero on 'SIP/224-09e1d728ZOMBIE' -- SIP/225-09e73388 answered Zap/1-1 Any ideas ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interfacing pri card to legacy pbx
I cannot tell for sure for any system, but we have an old Portmaster PM3 hooked-up from one port of our Sangoma A104d card, another one being from telco. So, yes you can emulate the telco from a sangoma A10x card. Here's what I have in my zapata.conf : ;Sangoma A104 port 1 [slot:12 bus:0 span: 1] switchtype=national pridialplan=unknown signalling=pri_cpe group=1 channel = 1-23 ;Sangoma A104 port 2 [slot:12 bus:0 span: 2] echocancel=no pridialplan=national signalling=pri_net group=2 channel = 25-47 You might have noticed that the signalling is different for both port. pri_net being the telco emulatin one. The clock needs also to be set on master in the wancfg utility. Another thing, you might want to consider using a 2 port card for that, because the clock master needs a reference and I can't tell for sure if it'll work with a reference from another card. Regards, Nicolas Hi guys, Can I use a Sangoma a101 to interface a legacy pbx to an Asterisk server? The pbx doesn't have sip and I want to come in off of a sip trunk and interface with the older system. I know I can use a pri card to hook in to the phone network, but can I use this same card to feed back the signaling as if I were the phone company to the older system? Thanks, Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on SLOW solid state disk
Hi ! We are running our asterisk from a transcend ts8gifd25. The whole system, including the OS fit in this 8 gig disk. If you don't do any recording of calls, you don't need that much of speed. We have 20 or so SIP phones, a PRI trough a quad-port sangoma card, one other port is a pass-trough to a dial-up RAS, another port is a point-to-point T1 data link to our office. To date, we've handle a handfull of simultaniously calls without any performance degradation. Regards, I'm looking at building up a standard asterisk system fanless/no moving parts. I found a cheap solid state disk (Transcend TS32GSSD25S-M), but it is SLOW...25mb/sec read 8mb/sec write. Has anyone tried a slow disk like this on asterisk? Will this delay voice prompts or screw up ast/linux in any interesting way? (I know there are linux distros and Asterisk projects designed to run off CF, but I'm hoping to stay mainstream) Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_sip Maximum retries exceeded on transmission
I have a situation here where a user has an AAstra 480i phone, which function corectly. The phone is behing a nat-router (a linksys wrv200 for it's VPN point to point facility). The phone is plugued in a port wich has qos enabled. And when the user places a call, sometimes (not always), we get this in the console : [May 8 13:41:55] WARNING[5804]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 1867350652 (Critical Response) [May 8 13:41:55] WARNING[5804]: chan_sip.c:1972 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. And then, the call stops. I did enabled sip debug for that peer, and I saw that a packet was sent to the phone, and the phone did reply (I see it in the debug). In sip.conf, for that peer, I have nat=yes. Any ideas ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS gateway recommendation
Here, we've tested in the past zimsms, but now the,re closed. We founded out that the most reliable and cost-effective way to send sms is with GPRS modem. Multitech manufacures excelent quality product for gprs, edge and others. So we opened up an sms-only acount, it costs us 10$ / month for 1000 sms sent, plus 10c each after, plus some other fees like 911 fees, etc. We generelay sent no more than a couple hundeds. With this modem hookup-up to a serial port, and a little program assuring the sending / receiving of sms (sms_link from Sourceforge)., and voilà ! Here in Canada, all cellular providers are hooked-up one to another for passing of sms', so I send out my sms' with Rogers, but any user in any network receives it in about 1 to 5 seconds. I don't know if it's the same in the US or other countries. If you have more questions about the specifics, you can contact me. Nicolas - Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 24, 2007 6:54 AM Subject: Re: [asterisk-users] SMS gateway recommendation Hash: SHA1 I personally use clickatell. Can you please comment on the reliability of clickatell, pricing and speed of delivery of the message to the end-user? But I use a PHP script to do the texting and call that from the dialplan. Is that script available somewhere? Thanks, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connection astrisk to a RAS (portmaster)
We've finaly solved it... For once, when we plugued the PRI into the first port, all began to work better. As for my dial-out into my RAS, I had to put pridialplan=unknown into my channels group of incoming line, and pridialplan=national in my group for my pri going to my RAS. That did the trick. Thanks, Nicolas - Original Message - From: Nicolas Ross [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 01, 2007 10:31 AM Subject: Re: [asterisk-users] Connection astrisk to a RAS (portmaster) Thanks, That's not it. I juste uninstalled wanpipe, redownloaded zaptel (another version (1.4.5.1) to try), re-installed wanpipe, (patching zaptel in the process), recompile wanpipe, re-compiled zaptel, recompiled asterisk, re-installed everything, and still got the same errors... Nicolas - Original Message - From: Jared Smith [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 01, 2007 9:15 AM Subject: Re: [asterisk-users] Connection astrisk to a RAS (portmaster) On Wed, 2007-10-31 at 20:57 -0400, Nicolas Ross wrote: I also get sometime : == Primary D-Channel on span 2 down [Oct 31 20:50:53] WARNING[10250]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 48 as D-channel anyway! == Primary D-Channel on span 2 up I don't claim to be an expert on the Wanpipe drivers, but it's been my experience that if your D-channels bounce up and down like that every few seconds, that the Wanpipe drivers didn't successfully patch Zaptel, or you haven't restarted Zaptel since the Wanpipe drivers patched Zaptel. Hopefully that's enough to get you up and running. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connection astrisk to a RAS (portmaster)
Thanks, That's not it. I juste uninstalled wanpipe, redownloaded zaptel (another version (1.4.5.1) to try), re-installed wanpipe, (patching zaptel in the process), recompile wanpipe, re-compiled zaptel, recompiled asterisk, re-installed everything, and still got the same errors... Nicolas - Original Message - From: Jared Smith [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 01, 2007 9:15 AM Subject: Re: [asterisk-users] Connection astrisk to a RAS (portmaster) On Wed, 2007-10-31 at 20:57 -0400, Nicolas Ross wrote: I also get sometime : == Primary D-Channel on span 2 down [Oct 31 20:50:53] WARNING[10250]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 48 as D-channel anyway! == Primary D-Channel on span 2 up I don't claim to be an expert on the Wanpipe drivers, but it's been my experience that if your D-channels bounce up and down like that every few seconds, that the Wanpipe drivers didn't successfully patch Zaptel, or you haven't restarted Zaptel since the Wanpipe drivers patched Zaptel. Hopefully that's enough to get you up and running. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connection astrisk to a RAS (portmaster)
Here's my planed setup :
PRI from telco -- (port 1 of A104d) * (port 2 of A104d) -- PM3
The PM3, for those who don't know is lucent's portmaster RAS dial-up router.
I had setup asterisk, zaptel, libpri, wanpipe (as I have sangoma's cards).
In wancfg, I have port 1 as TDM_VOICE, with hardware echo on, span 1. Port 2
is TDM_VOICE, without hw echo, Clock as master, reference 0 (for the time
being, I'm still not hooked up with my pri, will be 1 in the future), span
2. I alswo had to enable High Impedance on port 2 to operate without alarms.
Zaptel.conf:
loadzone=us
defaultzone=us
#Sangoma A104 port 1 [slot:12 bus:0 span: 1]
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
#Sangoma A104 port 2 [slot:12 bus:0 span: 2]
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48
zapata.conf (part of) :
;Sangoma A104 port 1 [slot:12 bus:0 span: 1]
switchtype=national
pridialplan=unknown
context=demo
group=1
signalling=pri_cpe
channel = 1-23
;Sangoma A104 port 2 [slot:12 bus:0 span: 2]
switchtype=national
pridialplan=unknown
context=demo
group=2
signalling=pri_net
channel = 25-47
Now, after start wanpipe and asterisk, my PM3 shows that the line is up (via
a t1 cross-over cable). I see that the channels are idle and waiting.
On * console, I get :
Primary D-Channel on span 2 up
All the time
I also get sometime :
== Primary D-Channel on span 2 down
[Oct 31 20:50:53] WARNING[10250]: chan_zap.c:2393 pri_find_dchan: No
D-channels available! Using Primary channel 48 as D-channel anyway!
== Primary D-Channel on span 2 up
[Oct 31 20:51:05] ERROR[10250]: chan_zap.c:8178 zt_pri_error: !! Got reject
for frame 1, but we only have others!
In my extensions.conf, I have :
exten = 1234567,1,Dial(Zap/g2/${EXTEN})
Whem I trie that extension via a softphone, I hear hald a ring, and nothing
else. The d-channel up continue to appear on the console.
Any help would be appriciated.
Nicolas
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