[asterisk-users] (no subject)
Generate $500 $2500 a month - Own Your Own Business http://parkovani-u-letiste-praha.cz/httpwagregerw2.php?aforcamp=329 Well, this is it, Capet. kevon wingate Wed, 16 May 2012 18:07:05 -- _ -- 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] Outgoing SIP calls dropped after 30 seconds.
Not using the CDR for billing, but I do use it to see usage and to know if it's cheaper to purchase a provider with unlimited incoming and pay-per-minute outgoing. I disabled 'SIP Transformation' in the SonicWall and so far so good (10/10 calls worked, more testing to be had, stay tuned.) On Sat, Nov 8, 2008 at 5:12 AM, Steve Totaro [EMAIL PROTECTED] wrote: Usually, calls terminating at 30 seconds is a sure sign that you need to add an Answer() in your dialplan. Try dropping that in before you dial out. I have seen this so many times and Answer() has always fixed the issue. The magic number is 30 seconds. Depending on if you use your CDRs for anything, especially billing, you may need to figure a way around that, since even if a call rings out, the CDR will reflect Answered. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Fri, Nov 7, 2008 at 10:38 PM, Grey Man [EMAIL PROTECTED] wrote: To get to the bottom of it I'd recommend determining why the ACKs are not getting through to Asterisk rather than trying to work around it. I'm actually suprised Asterisk terminates the call by default when it doesn't get the ACK to it's 200 Ok response that must be new for 1.4.22 as I haven't seen that behaviour in earlier versions. In my opinion it's unwarranted behaviour, if Asterisk is getting RTP then it should leave the call up irrespective of whether it gets an ACK or not. From the original SIP trace the ACK does not appear to be arriving at your Asterisk server at all. Try doing a packet trace on the network segment where the calling SIP agent is and see where it's trying to send the ACK to. My guess would be your firewall is incorrectly handling the SIP messages. Generally it's very bad news to use an ALG or firewall to mangle SIP packets as they almost always get it wrong. In your case there is a Record-Route header in the response so the ACK request should be being sent to that address. Perhaps your firewall is not correctly mangling that to allow the request to find its way back to your Asterisk server. Record-Route: sip:216.82.224.202;lr;ftag=as3ed791f3 Regards, Greyman. ___ -- 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 ___ -- 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] Outgoing SIP calls dropped after 30 seconds.
Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Problem: Make a call on a Polycom 320 IP phone to any number and (4/5 times) it will drop the call after 30 seconds. I noticed that the little timer that pops up on the LCD on the phone is missing when a call will be dropped. This timer appears when the phone is answered, so I have about 30 seconds to talk to them before the call is just dropped. Known Causes: It's a NAT issue, I know that much, I just don't know how to fix it. SIP debugging shows that it attempts to retransmit packets to my phone and since it can't, it drops it after 30 seconds. Log snippet: -- Executing [EMAIL PROTECTED]:19] Dial(SIP/203-b7a2b558, SIP/bw_outbound/+18005551212|300|) in new stack Audio is at public IP port 11968 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 216.82.224.202:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport From: 8881231234 sip:[EMAIL PROTECTED] IP;tag=as3ed791f3 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] IP Call-ID: [EMAIL PROTECTED] IP CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 07 Nov 2008 19:06:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 291 v=0 o=root 21520 21520 IN IP4 151.196.61.115 s=session c=IN IP4 public IP t=0 0 m=audio 11968 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -- Called bw_outbound/+18885551212 FreePBX*CLI --- SIP read from 216.82.224.202:5060 --- SIP/2.0 100 Giving a try Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060 From: 8881231234 sip:[EMAIL PROTECTED] IP;tag=as3ed791f3 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] IP CSeq: 102 INVITE Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 - --- (8 headers 0 lines) --- FreePBX*CLI --- SIP read from 216.82.224.202:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060 Record-Route: sip:216.82.224.202;lr;ftag=as3ed791f3 From: 8881231234 sip:[EMAIL PROTECTED] IP;tag=as3ed791f3 To: sip:[EMAIL PROTECTED];tag=VPST50603522629853 Call-ID: [EMAIL PROTECTED] IP CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Content-Type: application/sdp Content-Length: 184 v=0 o=- 1226084867 1226084868 IN IP4 209.244.42.253 s=- c=IN IP4 209.244.42.253 t=0 0 m=audio 64706 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 - --- (10 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.244.42.253:64706 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.244.42.253:64706 -- SIP/bw_outbound-08bf43d0 is making progress passing it to SIP/203-b7a2b558 Audio is at public IP port 16244 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (NAT) to 172.16.2.203:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.2.203;branch=z9hG4bKed293df65EAFD78F;received=172.16.2.203 From: Me sip:[EMAIL PROTECTED] IP;tag=28354B-27A53F00 To: sip:[EMAIL PROTECTED] IP;user=phone;tag=as600b952c Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] IP Content-Type: application/sdp Content-Length: 291 v=0 o=root 21520 21520 IN IP4 public IP s=session c=IN IP4 public IP t=0 0 m=audio 16244 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - --- (10 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.244.42.253:64706 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Re: [asterisk-users] Outgoing SIP calls dropped after 30 seconds.
Ok, recompiling it now with a 1 instead of XMIT_CRITICAL. Will check back to see if it worked. Would be nice if it did :) Thanks, Kurt On Fri, Nov 7, 2008 at 3:38 PM, Doug [EMAIL PROTECTED] wrote: At 14:15 11/7/2008, SIP wrote: Kurt Knudsen wrote: Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Question: Why does it sometimes work and sometimes not? This makes no sense and it happens on all phones. Any suggestions? We see this on occasion. It sounds a lot like Asterisk doing its usual routine of deciding that you can't POSSIBLY have a call going through because it can't receive an ACK response properly. Asterisk tries several times to send an ACK and get a response. If the remote system routes ACKs differently than it routes everything else, often times those ACKs get lost, and Asterisk assumes that the call can't be working, so it destroys it. ACK handling is a bit tricky in the real world, and we've run across countless incorrectly-configured SIP servers that don't handle it properly, so calls to them last just about exactly 30 seconds and then drop. There is, unfortunately, no way to turn off Asterisk's 'intelligent' behaviour in this scenario short of possibly patching the code. http://lists.digium.com/pipermail/asterisk-users/2007-May/187951.html ___ -- 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
Re: [asterisk-users] Outgoing SIP calls dropped after 30 seconds.
That seems to have sort of worked. It seems the phone decided to end the call this time, instead of Asterisk and now the call is dangling inside of 'sip show channels'. So that solution didn't work :( On Fri, Nov 7, 2008 at 4:28 PM, Kurt Knudsen [EMAIL PROTECTED] wrote: Ok, recompiling it now with a 1 instead of XMIT_CRITICAL. Will check back to see if it worked. Would be nice if it did :) Thanks, Kurt On Fri, Nov 7, 2008 at 3:38 PM, Doug [EMAIL PROTECTED] wrote: At 14:15 11/7/2008, SIP wrote: Kurt Knudsen wrote: Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Question: Why does it sometimes work and sometimes not? This makes no sense and it happens on all phones. Any suggestions? We see this on occasion. It sounds a lot like Asterisk doing its usual routine of deciding that you can't POSSIBLY have a call going through because it can't receive an ACK response properly. Asterisk tries several times to send an ACK and get a response. If the remote system routes ACKs differently than it routes everything else, often times those ACKs get lost, and Asterisk assumes that the call can't be working, so it destroys it. ACK handling is a bit tricky in the real world, and we've run across countless incorrectly-configured SIP servers that don't handle it properly, so calls to them last just about exactly 30 seconds and then drop. There is, unfortunately, no way to turn off Asterisk's 'intelligent' behaviour in this scenario short of possibly patching the code. http://lists.digium.com/pipermail/asterisk-users/2007-May/187951.html ___ -- 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] No incoming audio on Dahdi channels (TDM410P)
A previous issue has popped up and once again I'm out of ideas. During the evenings it seems that the TDM channels will spike (dahdi_monitor) and will refuse to listen for audio of any type, this includes DTMF. The only resolution I know of is to stop Asterisk and restart the dahdi service, but that's not a solution. All channels look like this, even the FXS. [EMAIL PROTECTED] Hardware]# dahdi_monitor 1 -vv Visual Audio Levels. Use chan_dahdi.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX (TX ###* Rx: 30076 (30076) Tx: 0 (0) I've stopped every service except SSH and networking (according to service --status-all) and nothing has changed. [EMAIL PROTECTED] cat /proc/interrupts CPU0 0: 77924086IO-APIC-edge timer 1: 3IO-APIC-edge i8042 6: 6IO-APIC-edge floppy 7: 0IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 1 IO-APIC-level acpi 12: 4IO-APIC-edge i8042 14: 104093IO-APIC-edge ide0 15: 690398IO-APIC-edge ide1 201: 77835719 IO-APIC-level wctdm24xxp0 209: 770795 IO-APIC-level eth1 NMI: 0 LOC: 77927794 ERR: 0 MIS: 0 Nothing looks shared, but then I see this in lspci -vb: 00:02.0 VGA compatible controller: Intel Corporation 82845G/GL[Brookdale-G]/GE Chipset Integrated Graphics Device (rev 03) (prog-if 00 [VGA controller]) Subsystem: Micro-Star International Co., Ltd. Unknown device 5578 Flags: bus master, fast devsel, latency 0, IRQ 11 Memory at d000 (32-bit, prefetchable) Memory at dff8 (32-bit, non-prefetchable) Capabilities: [d0] Power Management version 1 ... ... 01:01.0 Ethernet controller: Digium, Inc. Unknown device 8005 (rev 11) Subsystem: Digium, Inc. Unknown device 8005 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at cc00 Memory at dfdffc00 (32-bit, non-prefetchable) Expansion ROM at dfdc [disabled] Capabilities: [c0] Power Management version 2 Is that normal? Here's the output of dahdi_diag 1: dahdi: Dump of DAHDI Channel 1 (WCTDM/0/0,1,1): dahdi: flags: 201 hex, writechunk: ee0d008c, readchunk: ee0d0098 dahdi: rxgain: f8b8c480, txgain: f8b8c480, gainalloc: 0 dahdi: span: e9460054, sig: 2004 hex, sigcap: 6085 hex dahdi: inreadbuf: -1, outreadbuf: -1, inwritebuf: -1, outwritebuf: -1 dahdi: blocksize: 0, numbufs: 2, txbufpolicy: 0, txbufpolicy: 0 dahdi: txdisable: 0, rxdisable: 0, iomask: 0 dahdi: curzone: , tonezone: 0, curtone: , tonep: 0 dahdi: digitmode: 0, txdialbuf: , dialing: 0, aftdialtimer: 0, cadpos. 0 dahdi: confna: 0, confn: 0, confmode: 0, confmute: 0 dahdi: ec: , echocancel: 0, deflaw: 0, xlaw: f8b6f2a0 dahdi: echostate: 00, echotimer: 0, echolastupdate: 0 dahdi: itimer: 0, otimer: 0, ringdebtimer: 0 No idea what any of that means or how it's relevant. dmesg is full of interrupt misses and polarity reversals: ... wctdm24xxp0: Missed interrupt. Increasing latency to 18 ms in order to compensate. wctdm24xxp0: Missed interrupt. Increasing latency to 19 ms in order to compensate. 29794979 Polarity reversed (1 - -1) 29795839 Polarity reversed (-1 - 1) wctdm24xxp0: Missed interrupt. Increasing latency to 20 ms in order to compensate. wctdm24xxp0: Missed interrupt. Increasing latency to 21 ms in order to compensate. wctdm24xxp0: Missed interrupt. Increasing latency to 22 ms in order to compensate. 31595924 Polarity reversed (1 - -1) 31596867 Polarity reversed (-1 - 1) ... RING on 1/2! 74920374 Polarity reversed (-1 - 1) NO RING on 1/2! 74921961 Polarity reversed (1 - -1) RING on 1/2! NO RING on 1/2! NO BATTERY on 1/2! BATTERY on 1/2 (-)! Running AsteriskNow 1.5. X Windows is disabled. Ideas? Suggestions? Thoughts? Going to build another PC and toss this in there to see what happens tonight. 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
Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio
Any updates? It still seems to happen, though not as often as it used to. We're using Polycom 320 phones, if that makes a difference, though we did do it with X-Lite as well. On Sat, Oct 11, 2008 at 3:03 PM, Kurt Knudsen [EMAIL PROTECTED]wrote: Thanks, Steve, That's what I am unsure of. I don't know how to limit 1 call per trunk. If that's an easy thing to setup, I'd love to see it. On Fri, Oct 10, 2008 at 10:20 PM, Steve Totaro [EMAIL PROTECTED] wrote: Oh, I thought you had logic to count the calls on the trunk. You should limit each trunk to one call. This is the primary reason besides the email that basically said that customer support structure has been changed and anything beyond the Demarc would not be supported, I the Demarc is simply their boxen, so unless it is on their side, you will not get any helpful support from Bandwidth, plus they CCed over 500 people by address instead of setting up a group. http://www.bandwidth.com/content/support/?page=standardSupport I am with Junction and while a trunk is not unlimited as far as price for usage, the amount of trunks is unlimited (or at least as unlimited as it can be since nothing is really unlimited). They asked that I try not to go over one call per second for any real duration, and that I not hammer one LATA do to limited interconnects. The other thing was Junctions was very easy to sign up with, great support, and configuration was a breeze. As for Bandwidth, I think they are solid but due to recent changes and the fact that you must pay per channel, as well as the setup process, I decided they were not for me. I will take a second look at your sip.conf and extensions.conf later to see if something jumps out at me. I suspect since you are setting up two separate trunks with Bandwidth, you need to limit each trunk to one call, rather than two. Thanks, Steve Totaro On Fri, Oct 10, 2008 at 9:47 PM, Kurt Knudsen [EMAIL PROTECTED]wrote: externip messes up DTMF detection, and by messes up I mean it doesn't detect it at all. Setting nat=yes or nat=no didn't make a difference either. When the trunks are in use, the calls are fine, no dropped audio. It only happens when a 3rd call is made and there's no trunk available. Thanks :) On Fri, Oct 10, 2008 at 7:09 PM, Steve Totaro [EMAIL PROTECTED] wrote: You need to configure your box for nat settings, externip and other settings in sip.conf and set nat=yes instead of nat=no. One way audio is almost always a NAT issue and those are two glaring things that would cause problems. Thanks, Steve Totaro On Fri, Oct 10, 2008 at 6:32 PM, Kurt Knudsen [EMAIL PROTECTED]wrote: Hi Steve, It's behind a NAT/Firewall but SIP translation is enabled and removing it from behind the firewall did nothing, it still dropped calls. The calls connect and everything works, but it dies when all trunks are in use and someone else tries to call out. It seems like even though both channels are in use, it tries to connect to the 2nd trunk and thus kills the audio. Nothing strange came up in Wireshark or the firewall logs. Thanks. On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen [EMAIL PROTECTED] wrote: Hello, We have 2 SIP trunks from Bandwidth.com and if both are in use and someone tries to dial out, they cause another call to get one-way audio (the caller hears us, we cannot hear them). This happens 100% of the time and Bandwidth.com doesn't offer any support. I don't see any setting that tells Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm currently using, or attempting to use, groups to solve this problem, but sometimes it works, sometimes it doesn't. It breaks when a call goes out on a Queue, because it seems to add each phone to the group, which breaks my GotoIf() statement. Here's some relevant information: Users.conf (added by Asterisk-GUI) [trunk_2] provider = Bandwidth (SIP) ; GUI metadata context = DID_trunk_2 hasexten = no hasiax = no hassip = yes host = 216.82.224.202 registeriax = no registersip = no usecallerid = yes nat = no ;Testing trunkname = Bandwidth.com (Sip) ; GUI metadata username = secret = disallow = all allow = ulaw,alaw,g726 sip.conf [general] context = frombandwidth ;other variables, etc. ;Added according to Bandwidth.com's wiki entry. Changed to inband because we were having DTMF issues. [bandwidth.com_inbound] host=216.82.224.202 port=5060 type=peer disallow=all allow=ulaw dtmfmode=inband canreinvite=no reinvite=no context=frombandwidth nat=no [bandwidth.com_outbound] host=216.82.224.202 port=5060 type=peer disallow=all allow=ulaw dtmfmode=rfc2833 nat=no fromuser=11234567890 extensions.conf [globals] ;…irrelevant stuff trunk_1 = Dahdi/g1 trunk_2 = SIP/trunk_2 OUT_2 = SIP
Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio
I tried using GROUP(), here's a snippet from the first post.
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix
it added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
exten = _+1.,n,Set(DID=${EXTEN:2})
exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
;What we use to dialout. Try SIP trunks first, then Dahdi trunk as backup.
;This is where it breaks. I tried to make it so there can't be more
than 2 calls on SIP channels at once.
;Since it counts the phone as a channel, and adds it to the group, I
had to use 4.
[internalphones]
exten = _1NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _1NXXNXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} = 4]?100)
;If the group has 2 or more calls, do not dial.
exten = _1NXXNXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)})
exten =
_1NXXNXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)
exten = _1NXXNXX,100,Playback(all-circuits-busy-now)
exten = _1NXXNXX,101,congestion()
exten = _1NXXNXX,102,busy()
;This is where incoming calls go to if I'm awake.
[DID_trunk_2_timeinterval_Awake]
exten = _NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _NXXNXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})
exten = _NXXNXX,n,Set(CALLERID(num)=1${CALLERID(num)})
exten = _NXXNXX,n,Goto(voicemenu-custom-1|s|1)
I'll try playing around with incoming/outgoing and see if that makes a
difference. I don't know why it counts the phone as a channel, though.
On Mon, Oct 20, 2008 at 12:14 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
Tried using GROUP()?
When a call comes in or goes out:
Exten = XXX,1,Set(GROUP(bdwi_out_1)=outgoing/incoming);
Exten = XXX,n,GotoIf($[${GROUP_COUNT(outgoing/[EMAIL PROTECTED])}] 1?fail)
Exten = XXX,n,Dial(…)
Exten = XXX(fail),1,Congestion();
Exten = XXX(fail),n,Hangup();
Obviously choose outgoing or incoming, if you want to track both you can just
use $MATH() to add them together.
Or some other math logic to check the result.
On incoming Set(DIALSTATUS=CHANUNAVAIL) and it'll ring busy to bandwidth(or
out of service, you can tweak this).
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt Knudsen
Sent: Monday, October 20, 2008 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent
one-way audio
Any updates? It still seems to happen, though not as often as it used to.
We're using Polycom 320 phones, if that makes a difference, though we did do
it with X-Lite as well.
On Sat, Oct 11, 2008 at 3:03 PM, Kurt Knudsen [EMAIL PROTECTED] wrote:
Thanks, Steve,
That's what I am unsure of. I don't know how to limit 1 call per trunk. If
that's an easy thing to setup, I'd love to see it.
On Fri, Oct 10, 2008 at 10:20 PM, Steve Totaro [EMAIL PROTECTED] wrote:
Oh, I thought you had logic to count the calls on the trunk. You should
limit each trunk to one call. This is the primary reason besides the email
that basically said that customer support structure has been changed and
anything beyond the Demarc would not be supported, I the Demarc is simply
their boxen, so unless it is on their side, you will not get any helpful
support from Bandwidth, plus they CCed over 500 people by address instead of
setting up a group.
http://www.bandwidth.com/content/support/?page=standardSupport
I am with Junction and while a trunk is not unlimited as far as price for
usage, the amount of trunks is unlimited (or at least as unlimited as it can
be since nothing is really unlimited). They asked that I try not to go over
one call per second for any real duration, and that I not hammer one LATA do
to limited interconnects.
The other thing was Junctions was very easy to sign up with, great support,
and configuration was a breeze.
As for Bandwidth, I think they are solid but due to recent changes and the
fact that you must pay per channel, as well as the setup process, I decided
they were not for me.
I will take a second look at your sip.conf and extensions.conf later to see
if something jumps out at me. I suspect since you are setting up two
separate trunks with Bandwidth, you need to limit each trunk to one call,
rather than two.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 9:47 PM, Kurt Knudsen [EMAIL PROTECTED] wrote:
externip messes up DTMF detection, and by messes up I mean it doesn't detect
it at all. Setting nat=yes or nat=no didn't make a difference either.
When the trunks are in use, the calls are fine, no dropped audio. It only
happens when a 3rd call is made and there's no trunk available.
Thanks :)
On Fri, Oct 10, 2008 at 7:09 PM, Steve Totaro [EMAIL PROTECTED] wrote:
You need to configure your box for nat settings, externip
Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio
The GotoIf works, because it does failover sometimes, just not all the
time, I followed instructions from here:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf
And it seems to work in other areas that I use it in a similar way. I
only have the Set(GROUP()) when we are making outgoing calls on the
SIP trunk or when there's an incoming call on the SIP trunk. Anything
on Dahdi doesn't get included. I don't know how to tell my phones and
channels apart, I'm not trying to add the phones to the group, just
the channels. Can you paste some of your extensions.conf since you
also use Bandwidth.com?
Thanks.
On Mon, Oct 20, 2008 at 8:30 PM, [EMAIL PROTECTED] wrote:
-- Kurt Knudsen wrote :
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
currently using, or attempting to use, groups to solve this problem, but
sometimes it works, sometimes it doesn't. It breaks when a call goes out on
a Queue, because it seems to add each phone to the group, which breaks my
GotoIf() statement. Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI metadata
context = DID_trunk_2
hasexten = no
hasiax = no
hassip = yes
host = 216.82.224.202
registeriax = no
registersip = no
usecallerid = yes
nat = no ;Testing
trunkname = Bandwidth.com (Sip) ; GUI metadata
username =
secret =
disallow = all
allow = ulaw,alaw,g726
sip.conf
[general]
context = frombandwidth
;other variables, etc.
;Added according to Bandwidth.com's wiki entry. Changed to inband because we
were having DTMF issues.
[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=inband
canreinvite=no
reinvite=no
context=frombandwidth
nat=no
[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser=11234567890
extensions.conf
[globals]
;…irrelevant stuff
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
OUT_2 = SIP/bandwidth.com_outbound
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix it
added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
exten = _+1.,n,Set(DID=${EXTEN:2})
exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
;What we use to dialout. Try SIP trunks first, then Dahdi trunk as backup.
;This is where it breaks. I tried to make it so there can't be more than 2
calls on SIP channels at once.
;Since it counts the phone as a channel, and adds it to the group, I had to
use 4.
[internalphones]
exten = _1NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _1NXXNXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} = 4]?100) ;If the
group has 2 or more calls, do not dial.
exten = _1NXXNXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)})
exten =
_1NXXNXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)
exten = _1NXXNXX,100,Playback(all-circuits-busy-now)
exten = _1NXXNXX,101,congestion()
exten = _1NXXNXX,102,busy()
;This is where incoming calls go to if I'm awake.
[DID_trunk_2_timeinterval_Awake]
exten = _NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _NXXNXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})
exten = _NXXNXX,n,Set(CALLERID(num)=1${CALLERID(num)})
exten = _NXXNXX,n,Goto(voicemenu-custom-1|s|1)
Thanks.
--
This message was sent on behalf of [EMAIL PROTECTED] at openSubscriber.com
http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/10416933.html
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio
Well, when it fails over to the Dahdi trunk, it doesn't dial properly,
so I think I broke the macro. I will add the Set(GROUP()) stuff inside
of that macro-trunkdial-0.3 context and see if that helps. But it's
weird that I can't dial out. Here's a bit of the full log:
DEBUG[8221] app_macro.c: Executed application: Dial
VERBOSE[8221] logger.c: -- Executing
[EMAIL PROTECTED]:2] GotoIf(SIP/207-0a1b3590, 20
0 1-CONGESTION|1:1-out|1) in new stack
VERBOSE[8221] logger.c: -- Goto
(macro-trunkdial-failover-0.3,1-CONGESTION,1)
DEBUG[8221] app_macro.c: Executed application: Gotoif
VERBOSE[8221] logger.c: -- Executing
[EMAIL PROTECTED]:1] Dial(SIP/207-0a1b3590,
Dahdi/g1/18005551212) in new stack
DEBUG[8221] dsp.c: dsp busy pattern set to 500,500
DEBUG[8221] chan_dahdi.c: Dialing '18005551212'
DEBUG[8221] chan_dahdi.c: Deferring dialing...
VERBOSE[8221] logger.c: -- Called g1/18005551212
DEBUG[8221] chan_dahdi.c: Sent deferred digit string: T18005551212w
DEBUG[8221] chan_dahdi.c: Done dialing, but waiting for progress
detection before doing more...
VERBOSE[8221] logger.c: -- Hungup 'DAHDI/1-1'
Not sure how it broke, but it won't use the Dahdi channel :( It just
goes to a busy signal after you dial. I tested on an analog phone and
it can dial out normally, so it's the system.
Thanks.
On Mon, Oct 20, 2008 at 2:29 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have a macro to dial out, similar to yours in that it fails over to
Zap/Dahdi trunks in the event our bandwidth stuff is overloaded.
I run this in a macro, and only set and check groups within that macro. I'm
confused why yours would attach to phones in any way, unless you mean phone
to phone calls, in that case don't set the group?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt Knudsen
Sent: Monday, October 20, 2008 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent
one-way audio
The GotoIf works, because it does failover sometimes, just not all the
time, I followed instructions from here:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf
And it seems to work in other areas that I use it in a similar way. I
only have the Set(GROUP()) when we are making outgoing calls on the
SIP trunk or when there's an incoming call on the SIP trunk. Anything
on Dahdi doesn't get included. I don't know how to tell my phones and
channels apart, I'm not trying to add the phones to the group, just
the channels. Can you paste some of your extensions.conf since you
also use Bandwidth.com?
Thanks.
On Mon, Oct 20, 2008 at 8:30 PM, [EMAIL PROTECTED] wrote:
-- Kurt Knudsen wrote :
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
currently using, or attempting to use, groups to solve this problem, but
sometimes it works, sometimes it doesn't. It breaks when a call goes out on
a Queue, because it seems to add each phone to the group, which breaks my
GotoIf() statement. Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI metadata
context = DID_trunk_2
hasexten = no
hasiax = no
hassip = yes
host = 216.82.224.202
registeriax = no
registersip = no
usecallerid = yes
nat = no ;Testing
trunkname = Bandwidth.com (Sip) ; GUI metadata
username =
secret =
disallow = all
allow = ulaw,alaw,g726
sip.conf
[general]
context = frombandwidth
;other variables, etc.
;Added according to Bandwidth.com's wiki entry. Changed to inband because we
were having DTMF issues.
[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=inband
canreinvite=no
reinvite=no
context=frombandwidth
nat=no
[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser=11234567890
extensions.conf
[globals]
;...irrelevant stuff
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
OUT_2 = SIP/bandwidth.com_outbound
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix it
added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
exten = _+1.,n,Set(DID=${EXTEN:2})
exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
;What we use to dialout. Try SIP trunks first, then Dahdi trunk as backup.
;This is where it breaks. I tried to make it so there can't be more than 2
[asterisk-users] TDM410P with EC doesn't detect DTMF after being on for ~1 hour
Now that I have a new card and my echo problems are 'mostly' solved, I have another major issue to deal with. After about an hour or so the card will stop detecting DTMF tones on incoming calls. dahdi_monitor shows the following: [EMAIL PROTECTED] wctdm24xxp]# dahdi_monitor 1 -v Visual Audio Levels. Use chan_dahdi.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX (TX ###* The channel is spiked and I need to stop asterisk and restart dahdi. Here's what the full log shows when it sees an incoming call: [Oct 20 18:49:38] VERBOSE[10629] logger.c: -- Starting simple switch on 'DAHDI/1-1' [Oct 20 18:49:39] NOTICE[10629] chan_dahdi.c: Got event 17 (Polarity Reversal)... [Oct 20 18:49:42] NOTICE[10629] chan_dahdi.c: Got event 18 (Ring Begin)... [Oct 20 18:49:44] NOTICE[10629] chan_dahdi.c: Got event 2 (Ring/Answered)... [Oct 20 18:49:44] VERBOSE[10629] logger.c: -- Executing [EMAIL PROTECTED]:1] ExecIf(DAHDI/1-1, 1|SetCallerPres|unavailable) in new stack [Oct 20 18:49:44] VERBOSE[10629] logger.c: -- Executing [EMAIL PROTECTED]:2] ExecIf(DAHDI/1-1, 1|Set|CALLERID(all)=unknown 000) in new stack The 3 events are always there when DTMF is ignored/not detected. Here's what the log shows with a correct call: [Oct 20 18:37:16] DEBUG[10563] dsp.c: dsp busy pattern set to 500,500 [Oct 20 18:37:16] VERBOSE[10611] logger.c: -- Starting simple switch on 'DAHDI/1-1' [Oct 20 18:37:17] VERBOSE[10611] logger.c: -- Executing [EMAIL PROTECTED]:1] ExecIf(DAHDI/1-1, 0|SetCallerPres|unavailable) in new stack [Oct 20 18:37:17] VERBOSE[10611] logger.c: -- Executing [EMAIL PROTECTED]:2] ExecIf(DAHDI/1-1, 0|Set|CALLERID(all)=unknown 000) in new stack [Oct 20 18:37:17] VERBOSE[10611] logger.c: -- Executing [EMAIL PROTECTED]:3] Goto(DAHDI/1-1, voicemenu-custom-3|s|1) in new stack [Oct 20 18:37:17] VERBOSE[10611] logger.c: -- Goto (voicemenu-custom-3,s,1) [Oct 20 18:37:17] VERBOSE[10611] logger.c: -- Executing [EMAIL PROTECTED]:2] Wait(DAHDI/1-1, 2) in new stack [Oct 20 18:37:17] DEBUG[10611] chan_dahdi.c: Ignore switch to REVERSED Polarity on channel 1, state 4 [Oct 20 18:37:17] DEBUG[10611] chan_dahdi.c: Ignoring Polarity switch to IDLE on channel 1, state 4 [Oct 20 18:37:17] DEBUG[10611] chan_dahdi.c: Polarity Reversal event occured - DEBUG 2: channel 1, state 4, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= 47$ [Oct 20 18:37:19] VERBOSE[10611] logger.c: -- Executing [EMAIL PROTECTED]:3] Set(DAHDI/1-1, TIMEOUT(digit)=2) in new stack [Oct 20 18:37:19] VERBOSE[10611] logger.c: -- Digit timeout set to 2 The events are ignored and the call goes through as it should. Also, when the call FAILS, the caller ID does not work. Here's the last bit of dmesg: NO BATTERY on 1/1! BATTERY on 1/1 (+)! 26939263 Polarity reversed (1 - -1) NO BATTERY on 1/1! 26940073 Polarity reversed (-1 - 1) BATTERY on 1/1 (+)! RING on 1/1! 26984808 Polarity reversed (1 - -1) NO RING on 1/1! 26986380 Polarity reversed (-1 - 1) NO BATTERY on 1/1! BATTERY on 1/1 (+)! I have no idea what that means (module is running with debug=1). 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
[asterisk-users] Unknown call every 30 minutes on the dot.
Here's some freaky stuff coming from Areski CDR tool: 101. 2008-10-13 03:41:23 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:20 102. 2008-10-13 03:11:30 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:21 103. 2008-10-13 02:41:23 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:21 104. 2008-10-13 02:11:22 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:21 105. 2008-10-13 01:41:21 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:21 106. 2008-10-13 01:11:21 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:21 107. 2008-10-13 00:41:29 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:21 108. 2008-10-13 00:11:21 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:21 When Asterisk see an incoming call without a caller ID, it sets it to unknown and 000. As you can see from the list above, it happens every 30 minutes almost to the second. It is still happening right now, unless that line is in use, in which case it'll try again 30 minutes later. I did notice this in the /var/log/asterisk/full log: [Oct 13 03:11:30] NOTICE[4243] chan_dahdi.c: Got event 17 (Polarity Reversal)... [Oct 13 03:11:38] WARNING[4243] chan_dahdi.c: CallerID returned with error on channel 'DAHDI/1-1' Normally, it says: [Oct 12 13:23:59] DEBUG[30124] chan_dahdi.c: Ignore switch to REVERSED Polarity on channel 1, state 4 [Oct 12 13:23:59] DEBUG[30124] chan_dahdi.c: Ignoring Polarity switch to IDLE on channel 1, state 4 [Oct 12 13:23:59] DEBUG[30124] chan_dahdi.c: Polarity Reversal event occured - DEBUG 2: channel 1, state 4, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= = -233440198 Any clues? TDM410P with 2 FXO ports and EC module. Running Fedora Core 9 in init:3 with USB disabled (to prevent IRQ conflicts). ___ -- 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] Text messaging and Asterisk
I use the 'generic' file in Postfix to map an email address that is not in use to someone's text messaging address. It'd be [EMAIL PROTECTED] ie: [EMAIL PROTECTED] Then, any email that gets sent to [EMAIL PROTECTED], will get automatically sent to that person's phone. On Mon, Oct 13, 2008 at 3:14 PM, Eric Chamberlain [EMAIL PROTECTED] wrote: On Oct 13, 2008, at 11:28 AM, C. Savinovich wrote: I mean is if someone know of an sms server or service that allows me to send outgoing text messaging. Are you sending SMS to known users or to any mobile phone user? If you are sending to a fixed user base, track down the email to SMS gateways for their carriers. Then sending an SMS is no different than sending an e-mail. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- 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
Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio
Thanks, Steve,
That's what I am unsure of. I don't know how to limit 1 call per trunk. If
that's an easy thing to setup, I'd love to see it.
On Fri, Oct 10, 2008 at 10:20 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
Oh, I thought you had logic to count the calls on the trunk. You should
limit each trunk to one call. This is the primary reason besides the email
that basically said that customer support structure has been changed and
anything beyond the Demarc would not be supported, I the Demarc is simply
their boxen, so unless it is on their side, you will not get any helpful
support from Bandwidth, plus they CCed over 500 people by address instead of
setting up a group.
http://www.bandwidth.com/content/support/?page=standardSupport
I am with Junction and while a trunk is not unlimited as far as price for
usage, the amount of trunks is unlimited (or at least as unlimited as it can
be since nothing is really unlimited). They asked that I try not to go over
one call per second for any real duration, and that I not hammer one LATA do
to limited interconnects.
The other thing was Junctions was very easy to sign up with, great support,
and configuration was a breeze.
As for Bandwidth, I think they are solid but due to recent changes and the
fact that you must pay per channel, as well as the setup process, I decided
they were not for me.
I will take a second look at your sip.conf and extensions.conf later to see
if something jumps out at me. I suspect since you are setting up two
separate trunks with Bandwidth, you need to limit each trunk to one call,
rather than two.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 9:47 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
externip messes up DTMF detection, and by messes up I mean it doesn't
detect it at all. Setting nat=yes or nat=no didn't make a difference either.
When the trunks are in use, the calls are fine, no dropped audio. It only
happens when a 3rd call is made and there's no trunk available.
Thanks :)
On Fri, Oct 10, 2008 at 7:09 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
You need to configure your box for nat settings, externip and other
settings in sip.conf and set nat=yes instead of nat=no.
One way audio is almost always a NAT issue and those are two glaring
things that would cause problems.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 6:32 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hi Steve,
It's behind a NAT/Firewall but SIP translation is enabled and removing
it from behind the firewall did nothing, it still dropped calls. The calls
connect and everything works, but it dies when all trunks are in use and
someone else tries to call out. It seems like even though both channels are
in use, it tries to connect to the 2nd trunk and thus kills the audio.
Nothing strange came up in Wireshark or the firewall logs.
Thanks.
On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and
someone tries to dial out, they cause another call to get one-way audio
(the
caller hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that
tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
currently using, or attempting to use, groups to solve this problem, but
sometimes it works, sometimes it doesn't. It breaks when a call goes out
on
a Queue, because it seems to add each phone to the group, which breaks my
GotoIf() statement. Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI metadata
context = DID_trunk_2
hasexten = no
hasiax = no
hassip = yes
host = 216.82.224.202
registeriax = no
registersip = no
usecallerid = yes
nat = no ;Testing
trunkname = Bandwidth.com (Sip) ; GUI metadata
username =
secret =
disallow = all
allow = ulaw,alaw,g726
sip.conf
[general]
context = frombandwidth
;other variables, etc.
;Added according to Bandwidth.com's wiki entry. Changed to inband
because we were having DTMF issues.
[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=inband
canreinvite=no
reinvite=no
context=frombandwidth
nat=no
[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser=11234567890
extensions.conf
[globals]
;…irrelevant stuff
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
OUT_2 = SIP/bandwidth.com_outbound
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix
it added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP
[asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
currently using, or attempting to use, groups to solve this problem, but
sometimes it works, sometimes it doesn't. It breaks when a call goes out on
a Queue, because it seems to add each phone to the group, which breaks my
GotoIf() statement. Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI metadata
context = DID_trunk_2
hasexten = no
hasiax = no
hassip = yes
host = 216.82.224.202
registeriax = no
registersip = no
usecallerid = yes
nat = no ;Testing
trunkname = Bandwidth.com (Sip) ; GUI metadata
username =
secret =
disallow = all
allow = ulaw,alaw,g726
sip.conf
[general]
context = frombandwidth
;other variables, etc.
;Added according to Bandwidth.com's wiki entry. Changed to inband because we
were having DTMF issues.
[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=inband
canreinvite=no
reinvite=no
context=frombandwidth
nat=no
[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser=11234567890
extensions.conf
[globals]
;…irrelevant stuff
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
OUT_2 = SIP/bandwidth.com_outbound
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix it
added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
exten = _+1.,n,Set(DID=${EXTEN:2})
exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
;What we use to dialout. Try SIP trunks first, then Dahdi trunk as backup.
;This is where it breaks. I tried to make it so there can't be more than 2
calls on SIP channels at once.
;Since it counts the phone as a channel, and adds it to the group, I had to
use 4.
[internalphones]
exten = _1NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _1NXXNXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} = 4]?100) ;If the
group has 2 or more calls, do not dial.
exten = _1NXXNXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)})
exten =
_1NXXNXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)
exten = _1NXXNXX,100,Playback(all-circuits-busy-now)
exten = _1NXXNXX,101,congestion()
exten = _1NXXNXX,102,busy()
;This is where incoming calls go to if I'm awake.
[DID_trunk_2_timeinterval_Awake]
exten = _NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _NXXNXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})
exten = _NXXNXX,n,Set(CALLERID(num)=1${CALLERID(num)})
exten = _NXXNXX,n,Goto(voicemenu-custom-1|s|1)
Thanks.
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Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio
Hi Steve,
It's behind a NAT/Firewall but SIP translation is enabled and removing it
from behind the firewall did nothing, it still dropped calls. The calls
connect and everything works, but it dies when all trunks are in use and
someone else tries to call out. It seems like even though both channels are
in use, it tries to connect to the 2nd trunk and thus kills the audio.
Nothing strange came up in Wireshark or the firewall logs.
Thanks.
On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
currently using, or attempting to use, groups to solve this problem, but
sometimes it works, sometimes it doesn't. It breaks when a call goes out on
a Queue, because it seems to add each phone to the group, which breaks my
GotoIf() statement. Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI metadata
context = DID_trunk_2
hasexten = no
hasiax = no
hassip = yes
host = 216.82.224.202
registeriax = no
registersip = no
usecallerid = yes
nat = no ;Testing
trunkname = Bandwidth.com (Sip) ; GUI metadata
username =
secret =
disallow = all
allow = ulaw,alaw,g726
sip.conf
[general]
context = frombandwidth
;other variables, etc.
;Added according to Bandwidth.com's wiki entry. Changed to inband because
we were having DTMF issues.
[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=inband
canreinvite=no
reinvite=no
context=frombandwidth
nat=no
[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser=11234567890
extensions.conf
[globals]
;…irrelevant stuff
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
OUT_2 = SIP/bandwidth.com_outbound
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix it
added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
exten = _+1.,n,Set(DID=${EXTEN:2})
exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
;What we use to dialout. Try SIP trunks first, then Dahdi trunk as backup.
;This is where it breaks. I tried to make it so there can't be more than 2
calls on SIP channels at once.
;Since it counts the phone as a channel, and adds it to the group, I had
to use 4.
[internalphones]
exten = _1NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _1NXXNXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} = 4]?100) ;If
the group has 2 or more calls, do not dial.
exten = _1NXXNXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)})
exten =
_1NXXNXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)
exten = _1NXXNXX,100,Playback(all-circuits-busy-now)
exten = _1NXXNXX,101,congestion()
exten = _1NXXNXX,102,busy()
;This is where incoming calls go to if I'm awake.
[DID_trunk_2_timeinterval_Awake]
exten = _NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _NXXNXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})
exten = _NXXNXX,n,Set(CALLERID(num)=1${CALLERID(num)})
exten = _NXXNXX,n,Goto(voicemenu-custom-1|s|1)
Thanks.
http://lists.digium.com/mailman/listinfo/asterisk-users
Is your Asterisk box on a public IP or behind a NAT/Firewall?
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio
externip messes up DTMF detection, and by messes up I mean it doesn't detect
it at all. Setting nat=yes or nat=no didn't make a difference either.
When the trunks are in use, the calls are fine, no dropped audio. It only
happens when a 3rd call is made and there's no trunk available.
Thanks :)
On Fri, Oct 10, 2008 at 7:09 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
You need to configure your box for nat settings, externip and other
settings in sip.conf and set nat=yes instead of nat=no.
One way audio is almost always a NAT issue and those are two glaring things
that would cause problems.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 6:32 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hi Steve,
It's behind a NAT/Firewall but SIP translation is enabled and removing it
from behind the firewall did nothing, it still dropped calls. The calls
connect and everything works, but it dies when all trunks are in use and
someone else tries to call out. It seems like even though both channels are
in use, it tries to connect to the 2nd trunk and thus kills the audio.
Nothing strange came up in Wireshark or the firewall logs.
Thanks.
On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and
someone tries to dial out, they cause another call to get one-way audio
(the
caller hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
currently using, or attempting to use, groups to solve this problem, but
sometimes it works, sometimes it doesn't. It breaks when a call goes out on
a Queue, because it seems to add each phone to the group, which breaks my
GotoIf() statement. Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI metadata
context = DID_trunk_2
hasexten = no
hasiax = no
hassip = yes
host = 216.82.224.202
registeriax = no
registersip = no
usecallerid = yes
nat = no ;Testing
trunkname = Bandwidth.com (Sip) ; GUI metadata
username =
secret =
disallow = all
allow = ulaw,alaw,g726
sip.conf
[general]
context = frombandwidth
;other variables, etc.
;Added according to Bandwidth.com's wiki entry. Changed to inband
because we were having DTMF issues.
[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=inband
canreinvite=no
reinvite=no
context=frombandwidth
nat=no
[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser=11234567890
extensions.conf
[globals]
;…irrelevant stuff
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
OUT_2 = SIP/bandwidth.com_outbound
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix
it added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
exten = _+1.,n,Set(DID=${EXTEN:2})
exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
;What we use to dialout. Try SIP trunks first, then Dahdi trunk as
backup.
;This is where it breaks. I tried to make it so there can't be more than
2 calls on SIP channels at once.
;Since it counts the phone as a channel, and adds it to the group, I had
to use 4.
[internalphones]
exten = _1NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _1NXXNXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} = 4]?100) ;If
the group has 2 or more calls, do not dial.
exten = _1NXXNXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)})
exten =
_1NXXNXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)
exten = _1NXXNXX,100,Playback(all-circuits-busy-now)
exten = _1NXXNXX,101,congestion()
exten = _1NXXNXX,102,busy()
;This is where incoming calls go to if I'm awake.
[DID_trunk_2_timeinterval_Awake]
exten = _NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _NXXNXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})
exten = _NXXNXX,n,Set(CALLERID(num)=1${CALLERID(num)})
exten = _NXXNXX,n,Goto(voicemenu-custom-1|s|1)
Thanks.
http://lists.digium.com/mailman/listinfo/asterisk-users
Is your Asterisk box on a public IP or behind a NAT/Firewall?
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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[asterisk-users] How many outgoing phone line/voip account do I need?
Hi list, I have an application which has to automatically dial and send out a voice message to 50 different phone numbers at the same time. Does it mean that I need to sign up 50 phone lines or voip accounts in order to achieve this purpose? Is there a provider(voip prefer) who offer a special account which is able to handle multiple calls simultaneously? Thanks in advance. Kurt _ Find a local pizza place, movie theater, and more .then map the best route! http://maps.live.com/?icid=hmtag1FORM=MGAC01 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] trixbox web-administration
Hi list, trixbox web-administration can be reached by host ip. since I am trying trixbox on the machine where I host my website as well, can I move trixbox main page to xxx.xxx.xxx.xxx/asterisk? which file I should move and should I modify the file? Thanks. Kurt _ Get live scores and news about your team: Add the Live.com Football Page www.live.com/?addtemplate=footballicid=T001MSN30A0701 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio problems 50% of the time.
I have an Asterisk server that I use at work. I have a phone that is at home that logs into the Asterisk server at work. My home phone is hooked up via DSL through a Linksys router. You can see the my sip.conf for the phone blow. The problem is each time the phone rings I can hear/be heard 50% of the time. Any suggestion on what to look for. I do have my reg time set for 180 seconds on the cisco ATA186. [72459] type=friend username=XX secret=X host=dynamic context=voice-mail dtmfmode=rfc2833 ;canreivet=yes nat=yes qualify=yes Kurt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Audio problems 50% of the time.
I would agree, if I also experience choppy voice. Over the last month I had one spike of 893k over my T1. My average usually is 223k. I carved out 640k for voice QOS on the WAN router. At most I would have 4 calls up at once. The call comes in, the phone rings, 50% of the time I can have a conversations. 50% of time I can not. Maybe I should complain to my SIP service provider. Kurt --- if your connection is also used for web, email, and the worst, p2p, you better to have qos on your router. just be aware that g711 will use 80Kb up and down... gsm and g729 wil use 30/40Kb then : disallow all allow = gsm allow = g729 Olivier kurt x a écrit : I have an Asterisk server that I use at work. I have a phone that is at home that logs into the Asterisk server at work. My home phone is hooked up via DSL through a Linksys router. You can see the my sip.conf for the phone blow. The problem is each time the phone rings I can hear/be heard 50% of the time. Any suggestion on what to look for. I do have my reg time set for 180 seconds on the cisco ATA186. [72459] type=friend username=XX secret=X host=dynamic context=voice-mail dtmfmode=rfc2833 ;canreivet=yes nat=yes qualify=yes Kurt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 2620 as PRI gateway
A 488 can mean a codec miss match. Check that your Asterisk box is configured for g729. Kurt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 2620 as PRI gateway
debug ccsip message Kurt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mulitple voicemail on mulitple phones
I have four DIDs. 2400,2401,2402, and 2403 There is no phone attached to 2400 but the other three DIDs do have phones attached All the four DIDs have their own voicemail and voicemail works on all the DIDs. When you dial 2400 it rings the other three numbers. If no one picks up, it goes to the 2400 voicemail box. What I need to understand is how to notify the other three phones that voicemail was left on the 2400 extension. The other three DIDs must be able to access the 2400 voicemail, and delete it. Any ideas. Kurt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HDLC errors on PRI
Sounds like a timing issue or interop issue. Get rid of the NFAS (3rd t1 with all B channels) and make them all plain PRIs without D channel sharing. Jason Walker [EMAIL PROTECTED] wrote: I have looked through other postings to the user group for HDLC errors, went through what worked for other people, and still can not seem to get past this issue. For 3 days, I have been getting HDLC abort(6) errors in *. Prior to Tuesday, the circuits were clean...I had maybe 10 HDLC abort messages since August 10th. Here are my specs: 1 Gig IBM x300 w/ 1 Gig Ram 1 Quad TE405P card No errors on IRQs IRQs are separated with NO sharing hdparm for irq and dma are set to 'on' Software - FC1 with -1 updates to kernel, etc. Asterisk v 1.0.9, libpri 1.0.9, zaptel 1.0.9.2 1 T1 is a tieline to our Nortel Meridian 3 T1s are a PRI trunk group with D chans on 24 and 48. The third T1 only has b channels. No alarms from zttool. Calls go through, inbound and outbound. About every 5 seconds, I get the following on the console: Nov 4 21:10:37 NOTICE[9693]: PRI got event: Alarm (4) on Secondary D-channel of span 1Nov 4 21:10:37 NOTICE[9693]: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 The errors seem to increase as calls come in and out. There is also a noticable "popping" when the error happens. Any suggestions are welcome. thank you Jason ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! FareChase - Search multiple travel sites in one click. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice recognition
Does anyone know if Asterisk supports any Voice recognition software or is there a third party out that has one available for Asterisk. What I want to do with Voice recognition. When some calls my * IVR instead of the caller spelling the name via the buttons I want the user to be able to say the name. Kurt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Having Meetme call another conference
Is it possible to have a bunch of people call a meetme room then have that room call into another conference off net. T Kurt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Config PolyCom SoundStation 4000 help
I am trying to get a IP 4000 to register to Asterisk. I can make outbound calls from the IP 4000 but not to it. When I implement sip show peers it lists the extension but with no IP address (unspecified). I am configuring the phone via the web interface. I am not using ftp or tftp to configure the phone. Does anyone have a doc explaining how to get the phone to register to asterisk. Thanks, Kurt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk, cisco 3640's and DIDs...
Matching the Correct Inbound POTS Dial Peer for DID For DID to work correctly, make sure the incoming call matches the correct POTS dial-peer where the command direct-inward-dial is configured. If your PRI has DIDs you need the command. Kurt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial multiple phones
I need to able to ring 30 phones at once on * plus another 10 that are not on Asterisk. I know I can use the Dial(SIP/1SIP/2…SIP/30SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED]/109) but this seems cumbersome. Is there an easier way to do achieve this? Kurt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Queue ANI
I configured queues.conf and just added a bunch of member = SIP/ numbers to the bottom. I set up my extensions.conf with the access number to the queue. Everything works but the phones on the lists display a ANI of 911 out of area. Is there away to change that ANI to something else. Thanks Kurt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Packetization period for CODECs
Is it possible in * to set the Packetization period. For example: If I want G711 to be at 10ms. Is that possible in *? Thanks, Kurt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme Problem
I think I configured the MeetMe right. Since I am using SIP for inbound calls I followed the instruction, for 2.6 kernel, from this web page: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy When I call the MeetMe number I get the greeting to enter in your conference room. I do and get invalid conference room. Below is my configs: Executing Wait(SIP/192.168.1.2-08c82740, 1) in new stack -- Executing MeetMe(SIP/192.168.1.2-08c82740, ) in new stack -- Playing 'conf-getconfno' (language 'en') == Parsing '/etc/asterisk/meetme.conf': Found -- Playing 'conf-invalid' (language 'en') Sep 19 10:41:58 WARNING[14066]: file.c:554 ast_readaudio_callback: Failed sip.conf [15551232432] type=friend ;username=2432 ;secret=2432 host=dynamic context=voice-mail dtmfmode=rfc2833 ;canreivet=yes extension.conf [voice-mail] exten = _15551232432,1,wait(1) exten = _15551232432,2,Meetme exten = _15551232432,3,Hangup meetme.conf [voice-mail] conf = 100 Thanks Kurt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ztdummy configuration help
Upon setting up and configuring the my extension.conf, meetme.conf and following the instruction outlined at this web page: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy I get the following errors when calling the meetme number. Executing Wait(SIP/216.53.118.2-f41196e0, 1) in new stack -- Executing MeetMe(SIP/216.53.118.2-f41196e0, |sicp) in new stack -- Playing 'conf-getconfno' (language 'en') == Parsing '/etc/asterisk/meetme.conf': Found Sep 19 13:51:22 WARNING[14066]: chan_zap.c:757 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Sep 19 13:51:22 ERROR[14066]: chan_zap.c:6687 chandup: Unable to dup channel: No such file or directory Sep 19 13:51:22 WARNING[14066]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Sep 19 13:51:22 WARNING[14066]: app_meetme.c:230 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') Sep 19 13:51:23 WARNING[14066]: file.c:554 ast_readaudio_callback: Failed to write frame -- Playing 'conf-getconfno' (language 'en') Any help is greatly appreciated. Kurt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom Phone advise
I would like to know if any body is using the Polycom Soundstation IP 4000 SIP conference phone with Asterisk. I am thinking of purchasing one. Kurt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No translator path exists for channel type MGCP Comfort noise support incomplete
the fix is to disallow=g729 in mgcp.conf and to turn off silence supression in the ADIT600kurt turner [EMAIL PROTECTED] wrote: not trashing deb at all.. just wanted to see what would happen with redhat.. I'VE GIVEN UP and I'm reloading DEB.. I'm such a newb at this and I found more doc's with redhat support that's why I wanted to try the switch.. anyways HI HO hi ho back to Deb I go! know any doc put together for asterisk on debian sarge? especialy with h323 support?Michiel van Baak [EMAIL PROTECTED] wrote: On 09:38, Mon 15 Aug 05, kurt turner wrote: ONLY ON MONDAY! Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this? That's what you get from trashing Debian in favour ofRedHatPlease don't take this message seriously ;) Just couldn'tresist.Sorry-- Michiel van Baakhttp://michiel.vanbaak.info[EMAIL PROTECTED]GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D"Why is it drug addicts and computer afficionados are both called users?"___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Start your day with Yahoo! - make it your home page __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Comfort Noise incomplete - No translator path exists for channel type MGCP (native 4) to 256
I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC. I listed my conigs below
I go off hook and get this -
*CLI -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
*CLI
I dial 123 (an extension) and get this -
*CLI Aug 17 09:35:25 NOTICE[26024]: rtp.c:284 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.0.241 -- Executing Dial("MGCP/aaln/[EMAIL PROTECTED]", "MGCP/aaln/[EMAIL PROTECTED]") in new stackAug 17 09:35:28 WARNING[26024]: channel.c:2175 ast_request: No translator path exists for channel type MGCP (native 4) to 256Aug 17 09:35:28 NOTICE[26024]: app_dial.c:1091 dial_exec_full: Unable to create channel of type 'MGCP' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Congestion("MGCP/aaln/[EMAIL PROTECTED]", "") in new stack
*CLI
I hang up and get this -
*CLI == Spawn extension (outbound-default, 96017209841, 2) exited non-zero on 'MGCP/aaln/[EMAIL PROTECTED]'
here are the configs
:/etc/asterisk# more extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
[globals]
[extensions]
exten = 123,1,Dial,MGCP/aaln/[EMAIL PROTECTED]
exten = 124,1,Dial,MGCP/aaln/[EMAIL PROTECTED]
[directdial]
ignorepat = 9
exten = 9,1,MGCP/aaln/[EMAIL PROTECTED]
exten = 9,2,Congestion
[international]
ignorepat = 9
exten = _9011.,1,Dial(Zap/g2/${EXTEN:1})
exten = _9011.,2,Congestion
include = longdistance
[longdistance]
ignorepat = 9
exten = _91NXXNXX,1,Dial(Zap/g2/${EXTEN:1})
exten = _91NXXNXX,2,Congestion
include = local
[local]
ignorepat = 9
exten = _9NXXNXX,1,Dial,MGCP/aaln/[EMAIL PROTECTED]
exten = _9NXXNXX,2,Congestion
[outbound-default]
include = extensions
include = directdial
include = longdistance
include = local
IPD:/etc/asterisk#
***
:/etc/asterisk# more mgcp.conf
; MGCP Configuration for Asterisk
[general]
port=2727
bindaddr=0.0.0.0
allow=ulaw
allow=g729
allow=g726
tos=0x85
srvlookup=yes
wcardep=aaln/*
; Bob's CMG #1
[192.168.0.241]
context=outbound-default
host=192.168.0.241
wcardep=*
line = *
;
; Line 1
;
callerid = "John" 123
callgroup=0
pickupgroup=0
nat=no
threewaycalling=yes
transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer
callwaiting=yes ; this might be a cause of trouble for ip10s
cancallforward=yes
line = aaln/1
;
; Line 2
;
callerid = "Jane" 124
callgroup=0
pickupgroup=0
nat=no
threewaycalling=yes
transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer
callwaiting=yes ; this might be a cause of trouble for ip10s
cancallforward=yes
line = aaln/2
;outbound t1 port 5-6 cross connected to cmg 6:1:1:3-4
line = aaln/3
line = aaln/4
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[Asterisk-Users] No translator path exists for channel type MGCP Comfort noise support incomplete
ONLY ON MONDAY!
Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this?
*CLI -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: DownAug 15 11:20:19 NOTICE[13883]: rtp.c:284 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.0.241 -- Executing Dial("MGCP/aaln/[EMAIL PROTECTED]", "MGCP/aaln/[EMAIL PROTECTED]") in new stackAug 15 11:20:19 WARNING[13883]: channel.c:2175 ast_request: No translator path exists for channel type MGCP (native 4) to 256Aug 15 11:20:19 NOTICE[13883]: app_dial.c:1091 dial_exec_full: Unable to create channel of type 'MGCP' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'MGCP/aaln/[EMAIL PROTECTED]'
status
is 'CHANUNAVAIL'
*CLI
here is my mgcp.conf and extensions.conf
IPD:/etc/asterisk# more extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
[globals]
[extensions]
exten = 123,1,Dial,MGCP/aaln/[EMAIL PROTECTED]
exten = 124,1,Dial,MGCP/aaln/[EMAIL PROTECTED]
[directdial]
ignorepat = 9
exten = 9,1,MGCP/aaln/[EMAIL PROTECTED]
exten = 9,2,Congestion
[international]
ignorepat = 9
exten = _9011.,1,Dial(Zap/g2/${EXTEN:1})
exten = _9011.,2,Congestion
include = longdistance
[longdistance]
ignorepat = 9
exten = _91NXXNXX,1,Dial(Zap/g2/${EXTEN:1})
exten = _91NXXNXX,2,Congestion
include = local
[local]
ignorepat = 9
exten = _9NXXNXX,1,Dial,MGCP/aaln/[EMAIL PROTECTED]
exten = _9NXXNXX,2,Congestion
[outbound-default]
include = extensions
include = directdial
include = longdistance
include = local
and the *** mgcp.conf
IPD:/etc/asterisk# more mgcp.conf
; MGCP Configuration for Asterisk
[general]
port=2727
bindaddr=0.0.0.0
allow=ulaw
allow=g729
allow=g726
tos=0x85
srvlookup=yes
wcardep=aaln/*
; Bob's CMG #1
[192.168.0.241]
context=outbound-default
host=192.168.0.241
wcardep=*
line = *
;
; Line 1
;
callerid = "John" 123
callgroup=0
pickupgroup=0
nat=no
threewaycalling=yes
transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer
callwaiting=yes ; this might be a cause of trouble for ip10s
cancallforward=yes
line = aaln/1
;
; Line 2
;
callerid = "Jane" 124
callgroup=0
pickupgroup=0
nat=no
threewaycalling=yes
transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer
callwaiting=yes ; this might be a cause of trouble for ip10s
cancallforward=yes
line = aaln/2
;outbound t1 port 5-6 cross connected to cmg 6:1:1:3-4
line = aaln/3
line = aaln/4
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[Asterisk-Users] Comedian annoucment files
A user has their unavailable message played and once that message is over the Comedian message is played right after. Is there any way to prevent the Comedian message being played if the user's unavailable/busy message is being played. Thanks, Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I need a Asterisk tech
Hello, I am looking for a permanent part time tech that knows Asterisk, Digium FXO/E-1 cards and understands SIP and H.323 and general long distance telecom engineering. Please respond to me at my e-mail with your qualifications at: [EMAIL PROTECTED] or my number is 405-203-9162. Thank You Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 error when trying to start Asterisk
Asterisk has been working fine for me for several weeks using MGCP to a Adit600 for intra office calling. I have recently loaded h323 and the following errors occurs when starting asterisk. [chan_h323.so]Aug 10 09:09:18 WARNING[7824]: loader.c:258 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Aug 10 09:09:18 WARNING[7824]: loader.c:440 load_modules: Loading module chan_h323.so failed! Ouch ... error while writing audio data: : Broken pipe I loading the following : Open H.323 v1.12.2, PWLib v1.5.2, expat-dev-1.95, expat-1.95, openssl-devel-0.9.6b, openssl-0.9.6b - per the read me files I did try the advsie previously give here -- http://lists.digium.com/pipermail/asterisk-users/2003-April/011019.htmlbut that didn't work for me. I thinking I may have loaded these in the incorrect directories.. here is where they are located in (slash root) - is the following openh323 and pwlib located in root - /root/usr/src/asterisk/channels is the following - chan_h323.c - h323 - Would these being in different areas be the cause? Should I move these or remove them then reinstall them? Sorry for my noobness as I'm learning Linux for Asterisk.. I'm really a class 5 voice guy tryin to keep up!! Thanks, Kurt __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 error when trying to start Asterisk
yes.. i have the following IPD:/usr/local/lib# ls firmware libpt_linux_x86_d.so.1.5 libpt_linux_x86_r.so.1 python2.3libpt_linux_x86_d.so libpt_linux_x86_d.so.1.5.2 libpt_linux_x86_r.so.1.5libpt_linux_x86_d.so.1 libpt_linux_x86_r.so libpt_linux_x86_r.so.1.5.2 I found ldconfig under root /sbin/ldconfig when you say run ldconfig what are you saying? ldconfig -v .. right? if so I did that and I still get the h323 error listed below when firing up * anymore ideas?Derek Whitten [EMAIL PROTECTED] wrote: does libpt_linux_x86_r.so.1.5.2 exist on your machine?maybe try running ldconfig or if that file is in a non-standardlocation, maybe add that path to ld.so.conf and then run ldconfig againOn Wed, 2005-08-10 at 08:09, kurt turner wrote: Asterisk has been working fine for me for several weeks using MGCP to a Adit600 for intra office calling. I have recently loaded h323 and the following errors occurs when starting asterisk. [chan_h323.so]Aug 10 09:09:18 WARNING[7824]: loader.c:258 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directoryAug 10 09:09:18 WARNING[7824]: loader.c:440 load_modules: Loading module chan_h323.so failed! Ouch ... error while writing audio data: : Broken pipe I loading the following : Open H.323 v1.12.2, PWLib v1.5.2, expat-dev-1.95, expat-1.95, openssl-devel-0.9.6b, openssl-0.9.6b - per the read me files I did try the advsie previously give here -- http://lists.digium.com/pipermail/asterisk-users/2003-April/011019.html but that didn't work for me. I thinking I may have loaded these in the incorrect directories.. here is where they are located in (slash root) - is the following openh323 and pwlib located in root - /root/usr/src/asterisk/channels is the following - chan_h323.c - h323 - Would these being in different areas be the cause? Should I move these or remove them then reinstall them? Sorry for my noobness as I'm learning Linux for Asterisk.. I'm really a class 5 voice guy tryin to keep up!! Thanks, Kurt __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- -BEGIN GEEK CODE BLOCK-Version: 3.1GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w--PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK--___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail web access
My problems is when I log into the web page to get my voicemail I see that there nothing being listed. I know there is vmail their because I can retrieve the messages from the phone. I changed the following line in vmail.cgi so I do not need to login with my extension plus context. $context=local; # Define here your by default context (so you dont need to put [EMAIL PROTECTED] in the login I also created a new symbolic link to point to local direct instead to default: lrwxrwxrwx 1 root root 35 Jul 18 11:01 vm - /var/spool/asterisk/voicemail/local Any help would be greatly appreciated. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] vmail.cgi question
When I log into the web page to get my voicemail I see that there nothing being listed. I change the following line in vmail.cgi so I do not need to login with my extension plus context. $context=local; # Define here your by default context (so you dont need to put [EMAIL PROTECTED] in the login I also created a new symbolic link to point to local direct instead to default: lrwxrwxrwx 1 root root 35 Jul 18 11:01 vm - /var/spool/asterisk/voicemail/local Am I missing something else. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Timing out issue whenusing AGI
I have the below script that works but for one problem. The call
cannot last longer then 4 minutes when the script is utilized.
However, when I configure my extension.conf to not call the script the
call will stay up until I hang-up.
I call the script as follows:
exten = _24XX,1,AGI(internal.agi|${EXTEN})
exten = _24XX,2,hangup
A brief description of the script is that it allows my asterisk server
to route calls to two different PBXs.
It does not matter which PBX the call is sent to, it will always
hang-up after 4 minutes when using the script.
Any suggestions on what might cause this.
Kurt
#!/usr/bin/perl -w
use warnings;
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-answer(); #I tested with this command pounded and not pounded out.
my $val = $ARGV[0];
open(IN, /var/lib/asterisk/agi-bin/cme_db) or die $!;
my $ext = 0;
print STDERR $val\n;
while (IN) {
chomp;
$ext = $_;
# print STDERR $ext\n;
if ($ext == $val) {
$AGI-exec('Dial',SIP/$ext.'@cme-pbx');
close(IN);
goto EXIT;
}
} # end while loop
close(IN);
open(IN, /var/lib/asterisk/agi-bin/nortel_db) or die $!;
while (IN) {
chomp;
$ext = $_;
# print STDERR $ext\n;
if ($ext == $val) {
$AGI-exec('Dial',SIP/1555123$ext.'@nortel');
close(IN);
goto EXIT;
}
} # end while loop
close(IN);
$rc = $AGI-exec('Dial',SIP/$val.'|15|t');
if ($rc == 0) {
$AGI-exec('Voicemail',u$val);
goto EXIT;
}
EXIT: print STDERR Exiting Script\n;
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[Asterisk-Users] Asterisk Comedian Web page login
When I try to login into voicemail through the web interface It states incorrect login. In my voicemail.conf I have all voicemail boxes set under local. I changed the symbolic link to reflect the new directory under /var/spool/asterisk. Am I missing something? My vm link = /var/spool/asterisk/voicemail/local. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Early media dectection problem
I noticed when I call certain IVR systems, such as 1800calldhl, that Asterisk will not barge the prompt. Would this imply that Asterisk has an Early media detection problem. Is anyone else experiencing this problem. Is there a fix? Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial peer preference
Does Asterisk support preference for the dial peers. For example: I have two outbound peers in *. The first is a SIP dial peer and the second peer is to the PSTN via a T1. The SIP dial peer is the main outbound peer for all calls. However, if the my SIP providers network goes down, I need to be able to automatically route the call out the T1 card. Is this possible in Asterisk. I have not seen any preference commands for Asterisk. If not, is there a work around for this type of set up. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple Line config help
I have aSIPURA 841 that is working on L1 with phone # 4027. L2 is
configured for 94027.
Both numbers register with Asterisk. When issuing the command sip show peers
both numbers have the same IP address but 94027 show its sip port at
5061. Which I expect is right. When I dial 4027 it works but when I
dial 94027 I get a 486 busy here and voice mail picks up.
config below:
sip.conf
[4027]
type=friend
host=dynamic
dtmfmode=rfc2833
context=home
callerid=SIPURA-L1
canreinvite=no
[EMAIL PROTECTED]
[94027]
type=friend
host=dynamic
dtmfmode=rfc2833
context=home
callerid=SIPURA-L2
canreinvite=no
[EMAIL PROTECTED]
extensions.conf
exten = _40xx,1,Answer
exten = _40xx,2,Dial(SIP/${EXTEN},10,t)
exten = _40xx,3,Voicemail(u${EXTEN})
exten = _40xx,4,Hangup
exten = _40xx,103,Voicemail(b${EXTEN})
exten = _40xx,104,Hangup
exten = _940xx,1,Answer
exten = _940xx,2,Dial(SIP/${EXTEN},10,t)
exten = _940xx,3,Voicemail(u${EXTEN})
exten = _940xx,4,Hangup
exten = _940xx,103,Voicemail(b${EXTEN:1})
exten = _940xx,104,Hangup
Thanks,
Kurt
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Re: [Asterisk-Users] Multiple Line config help
That works. What I am tyring to do is have two separate DIDs. One is
4027 and the
other is 94207. Line 1 = DID 4027 and Line 2 = DID 94027. Dialing
4027 works to line
1 but dial 94027 gets a 486 busy.
Kurt
On 4/21/05, Henry Devito [EMAIL PROTECTED] wrote:
Don't you have to configure your dialplan to hunt to the next extensions?
How else would * know to try 94207 if 4207 is busy?
- Original Message -
From: kurt x [EMAIL PROTECTED]
To: Asterisk asterisk-users@lists.digium.com
Sent: Thursday, April 21, 2005 3:08 PM
Subject: [Asterisk-Users] Multiple Line config help
I have aSIPURA 841 that is working on L1 with phone # 4027. L2 is
configured for 94027.
Both numbers register with Asterisk. When issuing the command sip show
peers
both numbers have the same IP address but 94027 show its sip port at
5061. Which I expect is right. When I dial 4027 it works but when I
dial 94027 I get a 486 busy here and voice mail picks up.
config below:
sip.conf
[4027]
type=friend
host=dynamic
dtmfmode=rfc2833
context=home
callerid=SIPURA-L1
canreinvite=no
[EMAIL PROTECTED]
[94027]
type=friend
host=dynamic
dtmfmode=rfc2833
context=home
callerid=SIPURA-L2
canreinvite=no
[EMAIL PROTECTED]
extensions.conf
exten = _40xx,1,Answer
exten = _40xx,2,Dial(SIP/${EXTEN},10,t)
exten = _40xx,3,Voicemail(u${EXTEN})
exten = _40xx,4,Hangup
exten = _40xx,103,Voicemail(b${EXTEN})
exten = _40xx,104,Hangup
exten = _940xx,1,Answer
exten = _940xx,2,Dial(SIP/${EXTEN},10,t)
exten = _940xx,3,Voicemail(u${EXTEN})
exten = _940xx,4,Hangup
exten = _940xx,103,Voicemail(b${EXTEN:1})
exten = _940xx,104,Hangup
Thanks,
Kurt
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[Asterisk-Users] OutBOund Dial problem
I have the following extension (7700) that can dial out with the below config. exten = _1nxxnxx/7700,1,Dial(SIP/[EMAIL PROTECTED]) exten = _1nxxnxx/7700,2,Hangup If I change it to exten = _1nxxnxx/77XX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _1nxxnxx/77XX,2,Hangup It does not work. Any help is greatly appreciated. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXy dial tone problem
I have the Digium S100i IAXy device hooked up to my asterisk server. When I pick up the phone I do get dial tone but it does not stop when I start to dial a number. The dial tone is alway heard and it does not make the call. It does register with Asterisk I can make a call to the IAXy device and here ringing and voice in both direction. I did re-provision the device and reset the device a couple of times. The setup did work yesterday in both directions. The only difference between today and yesterday is the IP address but like I said I did re-provision and reset the device many times. The device is set up for DHCP. Any suggestions. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GotoIf problem
Removing the quotes and eliminating s,3,gotoif did work but its not
what I am looking for.
What I want to do is the following: If a ani that comes in has 10
digits I want to change the ${CALLERIDNUM} to unknown. If the ani
is 10 digits just goto voicemail.
When I set up my [vmail] to look like below, it does not work. When I
send a 4 digit
ani my e-mail confirmation of the voicemail shows the 4 digit ani and
not Unknown.
[globals]
Setvar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM})} != $DIGITS}]?4:5)
exten = s,4,Setvar(${CALLERIDNUM}=Unknown)
exten = s,5,Voicemail(u${ext})
exten = s,6,Hangup
Kurt
On Wed, 09 Mar 2005 07:34:24 +1100, Howard Lowndes [EMAIL PROTECTED] wrote:
On Wed, 2005-03-09 at 05:29, kurt x wrote:
I am trying to test how the GotoIf and $LEN functions work but am not
succeeding is
this venture. When I dial and access voicemail with an ani of 3000
the gotoif statement does not push the call to s|6. Its goes through
each line( 1,2,3,4,5,6,7) . In additon when I dial with a 10 digit
ani the s,3,Gotoif does not work. It also goes through each line(
1,2,3,4,5,6,7)
Any help is greatly appreciated.
Have you tried removing the quotes?
Thanks
Kurt
Asterisk CVS-HEAD-07/14/04-16:28:29 built by
[EMAIL PROTECTED] on a i686 running Linux
[globals]
${ext}=0
SetGlobalVar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM}}) = ${DIGITS}]?s|5)
exten = s,4,GotoIf($[${CALLERIDNUM} = 3000]?s|6)
exten = s,5,Voicemail(u${ext})
exten = s,6,Background(pbx-invalid)
exten = s,7,Hangup
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--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
--
When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft.
--
Flatter government, not fatter government;
Get rid of the Australian states.
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Re: [Asterisk-Users] GotoIf problem
To me it looks like the $LEN function is not working. When I do verbose start to * I see that it walks right through every step whether or not the ani is 10 digits or something else. Would it be better to write an AGI script? Kurt On Wed, 09 Mar 2005 11:41:50 -0600, Chris Wade [EMAIL PROTECTED] wrote: kurt x wrote: [globals] Setvar(DIGITS=10) Try this instead... [globals] DIGITS=10 -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GotoIf problem
I,ve gotten the GotoIf statement working now. I hard coded the value
10 in place of the ${DIGITS} varible. Worked like a charm.
Thanks to everyone who helped.
Kurt
On Wed, 09 Mar 2005 12:07:51 -0600, Chris Wade [EMAIL PROTECTED] wrote:
kurt x wrote:
[globals]
Setvar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM})} != $DIGITS}]?4:5)
exten = s,4,Setvar(${CALLERIDNUM}=Unknown)
exten = s,5,Voicemail(u${ext})
exten = s,6,Hangup
Oh, and it should be
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM})} != ${DIGITS}]?4:5)
-Chris
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[Asterisk-Users] GotoIf problem
I am trying to test how the GotoIf and $LEN functions work but am not
succeeding is
this venture. When I dial and access voicemail with an ani of 3000
the gotoif statement does not push the call to s|6. Its goes through
each line( 1,2,3,4,5,6,7) . In additon when I dial with a 10 digit
ani the s,3,Gotoif does not work. It also goes through each line(
1,2,3,4,5,6,7)
Any help is greatly appreciated.
Thanks
Kurt
Asterisk CVS-HEAD-07/14/04-16:28:29 built by
[EMAIL PROTECTED] on a i686 running Linux
[globals]
${ext}=0
SetGlobalVar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM}}) = ${DIGITS}]?s|5)
exten = s,4,GotoIf($[${CALLERIDNUM} = 3000]?s|6)
exten = s,5,Voicemail(u${ext})
exten = s,6,Background(pbx-invalid)
exten = s,7,Hangup
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Re: [Asterisk-Users] GotoIf problem
Will this do.
[general]
static=yes
writeprotect=no
[globals]
${ext}=0
SetGlobalVar(DIGITS=4)
[attendant]
;Main welcome message
exten = xxx2400,1,Answer()
exten = xxx2400,2,Wait(1)
exten = xxx2400,3,Background(welcome)
exten = xxx2400,4,ResponseTimeout,15
;Used for wrong button press
exten = i,1,Goto,invalid|s|1
;To reach the operator
exten = 0,1,Goto,operator|s|1
;Company directory seach feature
exten = 3,1,Directory(local|cme-pbx)
exten = 3,2,Hangup
;To access VoiceMailMain
exten = 9,1,Goto,voicemail|s|1
;Need to be able to dial 4 digit extensions
include = cme-pbx
[voice-mail]
include = attendant
exten = 3000,1,Answer
exten = 3000,2,Dial(SIP/3000)
exten = 3000,3,Hangup
;Used in conjuction with ResposeTimeout
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
;Number that CME dials to forward voice mail to Asterisk
exten = _22XXX,1,Setvar(ext=${EXTEN:1})
exten = _22XXX,2,Goto,vmail|s|1
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM})} = ${DIGITS}]?s|5)
exten = s,4,GotoIf($[${CALLERIDNUM} = 3000]?s|6)
exten = s,5,Voicemail(u${ext})
exten = s,6,Background(pbx-invalid)
exten = s,7,Hangup
;The below a option triggers on the button
; press *
exten = a,1,VoicemailMain
exten = a,2,Hangup
[cme-pbx]
exten = _24XX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _24XX,2,Hangup
[operator]
exten = s,1,Dial(SIP/[EMAIL PROTECTED])
[voicemail]
exten = s,1,VoicemailMain()
exten = s,2,Hangup
[invalid]
exten = s,1,Playback(pbx-invalid)
exten = s,2,Goto,attendant|xxx2400|3
On Tue, 08 Mar 2005 12:36:38 -0600, Dennis Webb [EMAIL PROTECTED] wrote:
Can you post your dialplan for that extension. Also, NoOp works great for
debugging these issues.
On Tue, 2005-03-08 at 12:29, kurt x wrote:
I am trying to test how the GotoIf and $LEN functions work but am not
succeeding is this venture. When I dial and access voicemail with an ani of
3000 the gotoif statement does not push the call to s|6. Its goes through
each line( 1,2,3,4,5,6,7) . In additon when I dial with a 10 digit ani the
s,3,Gotoif does not work. It also goes through each line( 1,2,3,4,5,6,7) Any
help is greatly appreciated. Thanks Kurt Asterisk CVS-HEAD-07/14/04-16:28:29
built by [EMAIL PROTECTED] on a i686 running Linux [globals]
${ext}=0 SetGlobalVar(DIGITS=10) [vmail] exten = s,1,Answer exten =
s,2,NoOp(${ext}) exten = s,3,GotoIf($[${LEN(${CALLERIDNUM}}) =
${DIGITS}]?s|5) exten = s,4,GotoIf($[${CALLERIDNUM} = 3000]?s|6) exten =
s,5,Voicemail(u${ext}) exten = s,6,Background(pbx-invalid) exten =
s,7,Hangup ___ Asterisk-Users
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[Asterisk-Users] IAX LAGRQ POKE explanation
Can someone explain in greater detail the following two Control frames. The IAX2 draft document had extremely brief explanations. LAGRQ = Lag request POKE = Poke request. Thanks, Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX trap question
I would like to know if the following lines represent the RTP traffic going across, the CODEC being used is G711ulaw, or both. The complete trap is below the dotted lines Thanks Kurt asterick*CLI Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 004 Type: VOICE Subclass: 4 Timestamp: 02570ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 02570ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 003 Type: CONTROL Subclass: (255?) Timestamp: 02778ms SCall: 1 DCall: 1 [192.168.2232:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 02778ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 005 ISeqno: 003 Type: VOICE Subclass: 4 Timestamp: 02780ms SCall: 1 DCall: 1 [192.168.2232:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 006 Type: IAX Subclass: ACK Timestamp: 02780ms SCall: 1 DCall: 1 [192.168.2232:4569] -- Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW asterick*CLI Timestamp: 4ms SCall: 1 DCall: 0 [192.168.2232:4569] VERSION : 2 CALLED NUMBER : 2001 CALLING NUMBER : 3000 LANGUAGE: en CALLED CONTEXT : home USERNAME: master asterick*CLI FORMAT : 4 CAPABILITY : 65287 ADSICPE : 2 DATE TIME : 174228915 asterick*CLI Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 9ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI AUTHMETHODS : 3 CHALLENGE : 759448742 USERNAME: master asterick*CLI Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 9ms SCall: 1 DCall: 1 [192.168.2232:4569] MD5 RESULT : 707018f7eb07cfa8a966853c868683a7 asterick*CLI Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00012ms SCall: 1 DCall: 1 [192.168.2232:4569] FORMAT : 4 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00012ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: CONTROL Subclass: ANSWER Timestamp: 00015ms SCall: 1 DCall: 1 [192.168.2232:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00015ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Mar 2 16:45:38 NOTICE[-1222640720]: rtp.c:285 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible asterick*CLI Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 002 Type: CONTROL Subclass: RINGING Timestamp: 00018ms SCall: 1 DCall: 1 [192.168.2232:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 00018ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 004 Type: VOICE Subclass: 4 Timestamp: 02570ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 02570ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 003 Type: CONTROL Subclass: (255?) Timestamp: 02778ms SCall: 1 DCall: 1 [192.168.2232:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 02778ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 005 ISeqno: 003 Type: VOICE Subclass: 4 Timestamp: 02780ms SCall: 1 DCall: 1 [192.168.2232:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 006 Type: IAX Subclass: ACK Timestamp: 02780ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 006 Type: IAX Subclass: HANGUP Timestamp: 08033ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 006 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 08033ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP phone speaker phone mic cutting out
Title: Message Hi i have an @home box with Uniden UIP200's and the speaker phone works great for recording my voicemail box and recording the digital receptionist menu but when i make calls out the PSTN the mic on the speaker phone acts like it only picks up when my voice is really high, make a noise and get closer to the mic all the sudden it starts working then if i back off it quits and visa-versa, but I know its not a problem with the mic because it works perfect between other phones in my office, just not when i make calls to the PSTN, if i use the handset it's fine too. Kurt Fankhauser WaveLincwww.wavelinc.com114 S. Walnut St.Bucyrus, OH 44820419-562-6405 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.5.0 - Release Date: 2/25/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip wifi phone?
Can't find em anywhere -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paradise Dove Sent: Monday, February 21, 2005 9:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sip wifi phone? what about senao SI-7800H? this is the link: http://www.senao.com.tw/english/product/product_wireless01_outdoor_1.asp ?pgtl=Wirelesstp1id=02tp2id=06proid=000131 On Mon, 21 Feb 2005 23:42:30 -0600, Kristian Kielhofner [EMAIL PROTECTED] wrote: Kurt Fankhauser wrote: Sounds like I'm going to have to wait and hope some new phones are released. Kurt, Check out my message from October: http://lists.digium.com/pipermail/asterisk-users/2004-October/067769.h tml and here is a link to the Broadcom page: http://www.broadcom.com/products/product.php?product_id=BCM1160catego ry_id=45 I really, really wish someone, anyone, would start cranking out some devices based off of these chips. Linksys is really into Broadcom. Why not them? (As long as they don't have a blue plastic case!) -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.3.0 - Release Date: 2/21/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 2/22/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX channel unable to create
I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by [EMAIL PROTECTED] on a i686 running Linux Box B is running: Asterisk 1.0.3 built by [EMAIL PROTECTED] on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12 WARNING[-1233155152]: channel.c:1860 ast_request: No channel type registered for 'IAX' Feb 21 12:48:12 NOTICE[-1233155152]: app_dial.c:696 dial_exec: Unable to create channel of type 'IAX' Feb 21 12:48:14 WARNING[-1116300368]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Non-critical Response) My box A iax.conf: [general] port=5036 bindport=5036 bandwidth=low allow=ulaw disallow=lpc10 jitterbuffer=no tos=lowdelay [slave] type=friend secret=4435 context=voice-mail defaultip=192.168.2.232 qualify=yes My Box A extension.conf [voice-mail] exten = _2000,1,Dial(IAX/master:[EMAIL PROTECTED]/[EMAIL PROTECTED]) My box B iax.conf [general] port=5036 bindport=5036 bandwidth=low allow=ulaw disallow=lpc10 tos=lowdelay [master] type=friend secret=4435 context=home defaultip=192.168.1.2 qualify=yes My Box B extension.conf [home] exten = _24xx,1,Dial(IAX2/slave:[EMAIL PROTECTED]/[EMAIL PROTECTED]) Thanks in advance Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip wifi phone?
Title: Message Does anyone know of any sip wifi phones? Only one i can find that is redily availiable is the zyxel prestige 2000w and from what i hear it is flaky. Kurt Fankhauser WaveLincwww.wavelinc.com114 S. Walnut St.Bucyrus, OH 44820419-562-6405 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.3.0 - Release Date: 2/21/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip wifi phone?
Sounds like I'm going to have to wait and hope some new phones are released. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Monday, February 21, 2005 7:55 PM To: Asterisk Users Subject: Re: [Asterisk-Users] sip wifi phone? Its not flaky at all. We have 2. The only bad thing is its lack of power. I'm not that too familiar with WiFi devices but it only has about 2hrs worth of talk time and about 10hrs of standby time. I'm not really sure on the standby time, but it had a full battery when I left it on my desk at 5 on fri; came back on Monday and it was dead. -Matthew From: Kurt Fankhauser [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 21 Feb 2005 20:34:18 -0800 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] sip wifi phone? Does anyone know of any sip wifi phones? Only one i can find that is redily availiable is the zyxel prestige 2000w and from what i hear it is flaky. Kurt Fankhauser WaveLinc HYPERLINK http://www.wavelinc.com/www.wavelinc.com 114 S. Walnut St. Bucyrus, OH 44820 419-562-6405 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.3.0 - Release Date: 2/21/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.3.0 - Release Date: 2/21/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.3.0 - Release Date: 2/21/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help with @home
Title: Message just reinstalled @home and i have a one of those 100 cards, anyways when i call from the pstn the box picks up but i hear nothing, then it clicks a couple times, then nothing again, i am trying to get the digital receptionist to work but it won't save my wav file to the @home box and all the radio buttons under incoming calls are greyed out. the greyed out thing seems to be my biggest problem right now, also do you have to use a ip phone to record your greeting because this wav file stuff isn't working. Kurt Fankhauser WaveLincwww.wavelinc.com114 S. Walnut St.Bucyrus, OH 44820419-562-6405 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] help with @home
Title: Message I'll buy a IP phone tomarrow so i can do that -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collinsSent: Sunday, February 20, 2005 2:40 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] help with @home Can you work through a process of elimination if you record the file using an internal extension by dialing *77 and seeing if that works? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt FankhauserSent: Sunday, February 20, 2005 7:42 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] help with @home just reinstalled @home and i have a one of those 100 cards, anyways when i call from the pstn the box picks up but i hear nothing, then it clicks a couple times, then nothing again, i am trying to get the digital receptionist to work but it won't save my wav file to the @home box and all the radio buttons under incoming calls are greyed out. the greyed out thing seems to be my biggest problem right now, also do you have to use a ip phone to record your greeting because this wav file stuff isn't working. Kurt Fankhauser WaveLincwww.wavelinc.com114 S. Walnut St.Bucyrus, OH 44820419-562-6405 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] help with @home
I think the box is answering calls but I don't think the digital receptionist is working properly. Kurt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Thompson Sent: Sunday, February 20, 2005 3:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] help with @home Kurt Fankhauser wrote: just reinstalled @home and i have a one of those 100 cards, anyways when i call from the pstn the box picks up but i hear nothing, then it clicks a couple times, then nothing again, i am trying to get the digital receptionist to work but it won't save my wav file to the @home box and all the radio buttons under incoming calls are greyed out. the greyed out thing seems to be my biggest problem right now, also do you have to use a ip phone to record your greeting because this wav file stuff isn't working. Are you logged into the console while your testing the dialing in? What messages are you seeing? If asterisk is already running in the background, do a asterisk -r before you start to dial in. If there is some other interface in the @home distribution for monitoring asterisk, you'll have to say what app you're using and what you're seeing. At any rate, without log, error, or console messages there's not alot we can do for you. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] help with @home
Title: Message i got the xlite phone on my pc and i have no idea how to get it working, do i have to add it into the * box or just change some settings on the softphone? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collinsSent: Sunday, February 20, 2005 3:36 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] help with @home Just download a free softphone and do it that way eg xten From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt FankhauserSent: Sunday, February 20, 2005 9:18 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] help with @home I'll buy a IP phone tomarrow so i can do that -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collinsSent: Sunday, February 20, 2005 2:40 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] help with @home Can you work through a process of elimination if you record the file using an internal extension by dialing *77 and seeing if that works? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt FankhauserSent: Sunday, February 20, 2005 7:42 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] help with @home just reinstalled @home and i have a one of those 100 cards, anyways when i call from the pstn the box picks up but i hear nothing, then it clicks a couple times, then nothing again, i am trying to get the digital receptionist to work but it won't save my wav file to the @home box and all the radio buttons under incoming calls are greyed out. the greyed out thing seems to be my biggest problem right now, also do you have to use a ip phone to record your greeting because this wav file stuff isn't working. Kurt Fankhauser WaveLincwww.wavelinc.com114 S. Walnut St.Bucyrus, OH 44820419-562-6405 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] help with @home
Title: Message i have got the softphone working, now i am trying to setup voicemail for my @home box, under extension i have voicemail directory enabled but when i call that extension it just keeps rining and never goes to voicemail kurt -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt FankhauserSent: Sunday, February 20, 2005 7:24 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] help with @home i got the xlite phone on my pc and i have no idea how to get it working, do i have to add it into the * box or just change some settings on the softphone? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collinsSent: Sunday, February 20, 2005 3:36 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] help with @home Just download a free softphone and do it that way eg xten From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt FankhauserSent: Sunday, February 20, 2005 9:18 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] help with @home I'll buy a IP phone tomarrow so i can do that -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collinsSent: Sunday, February 20, 2005 2:40 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] help with @home Can you work through a process of elimination if you record the file using an internal extension by dialing *77 and seeing if that works? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt FankhauserSent: Sunday, February 20, 2005 7:42 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] help with @home just reinstalled @home and i have a one of those 100 cards, anyways when i call from the pstn the box picks up but i hear nothing, then it clicks a couple times, then nothing again, i am trying to get the digital receptionist to work but it won't save my wav file to the @home box and all the radio buttons under incoming calls are greyed out. the greyed out thing seems to be my biggest problem right now, also do you have to use a ip phone to record your greeting because this wav file stuff isn't working. Kurt Fankhauser WaveLincwww.wavelinc.com114 S. Walnut St.Bucyrus, OH 44820419-562-6405 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] help with @home
Title: Message i got it -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt FankhauserSent: Sunday, February 20, 2005 8:01 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] help with @home i have got the softphone working, now i am trying to setup voicemail for my @home box, under extension i have voicemail directory enabled but when i call that extension it just keeps rining and never goes to voicemail kurt -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt FankhauserSent: Sunday, February 20, 2005 7:24 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] help with @home i got the xlite phone on my pc and i have no idea how to get it working, do i have to add it into the * box or just change some settings on the softphone? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collinsSent: Sunday, February 20, 2005 3:36 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] help with @home Just download a free softphone and do it that way eg xten From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt FankhauserSent: Sunday, February 20, 2005 9:18 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] help with @home I'll buy a IP phone tomarrow so i can do that -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collinsSent: Sunday, February 20, 2005 2:40 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] help with @home Can you work through a process of elimination if you record the file using an internal extension by dialing *77 and seeing if that works? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt FankhauserSent: Sunday, February 20, 2005 7:42 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] help with @home just reinstalled @home and i have a one of those 100 cards, anyways when i call from the pstn the box picks up but i hear nothing, then it clicks a couple times, then nothing again, i am trying to get the digital receptionist to work but it won't save my wav file to the @home box and all the radio buttons under incoming calls are greyed out. the greyed out thing seems to be my biggest problem right now, also do you have to use a ip phone to record your greeting because this wav file stuff isn't working. Kurt Fankhauser WaveLincwww.wavelinc.com114 S. Walnut St.Bucyrus, OH 44820419-562-6405 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Q.SIG support in CVS
--On February 18, 2005 11:16:11 -0600 [EMAIL PROTECTED] wrote: On Fri, Feb 18, 2005 at 02:18:37PM +0100, Kurt Bauer wrote: I just read thru the changelog.txt of the current CVS version and what catched my eye was the following line: 'Adding Q.SIG switchtype option to chan_zap' . But there is no sample config in zapata.conf for Q.SIG and no 'feature-list'. Does this exist anywhere or has anyone already has experience with * and Q.SIG and wants to share ?? Yeah, I've got some experience with it (I'm the one working on it :-) ). Right now we can do send/receive of DivertingLegInformation2 messages, message waiting indication activate/deactivate, and receive of calling name information. Oh, and of coure all your basic PRI stuff, such as call setup and teardown. So all I have to do is change the line 'switchtype=euroisdn' in zapata.conf to 'switchtype=Q.SIG' ??? I still haven't got the point how to tell my * Box that it is now Q.SIG aware :-o Thanks, br, Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk setup
Hi, I just joined the list, anyways i am trying to setup an @home box with a x100p card and so far i can't even get the box to pickup the incoming call and in the amp management under the section send calls from PSTN too page all the radio buttons are blank and i want to use the digital receptionist, also when i try to setup digital receptionist via uploading wav file and save, it says file uploaded successfully but when i go back in there nothing is in the digital receptionist page. Kurt Fankhauser WaveLinc www.wavelinc.com 114 S. Walnut St. Bucyrus, OH 44820 419-562-6405 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.8 - Release Date: 2/14/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Q.SIG support in CVS
Hi, I just read thru the changelog.txt of the current CVS version and what catched my eye was the following line: 'Adding Q.SIG switchtype option to chan_zap' . But there is no sample config in zapata.conf for Q.SIG and no 'feature-list'. Does this exist anywhere or has anyone already has experience with * and Q.SIG and wants to share ?? Thanks a lot in advance, best regards, Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceMail ANI question
When I receive voicemail notification via e-mail I noticed that the
${VM_CALLERID) puts the IP address of the * box when callee info is
not present. Is there a way to have the field put Unkown caller in
instead of the IP address of the * box.
Kurt
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[Asterisk-Users] Rining Issues
When I access the Directory() and use it to call an extension, the origination hears garbled or inconsistent ringing. The termination side rings normally and the conversation is clean in both directions. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Directory() ringing problem
The Directory command is working properly but the ringing herd in the origination phone is either garbled or herd infrequently. The termination phone does ring with consistency. Any suggestion on what might be happening. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IVR Timing out
I set up an IVR systems that plays a message for 15 seconds but once the message is over you can not select any of the prompts. If you select something within 10 seconds the IVR system works. I even set the ResponseTimeout cmd to 25 secs but that does not work. Jan 24 09:54:29 NOTICE[-1222644816]: sched.c:221 sched_settime: Request to schedule in the past?!?! [attendant] ;Main welcome message exten = s,1,Wait(2) exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,25 exten = s,4,Background(welcome_n2p1) exten = s,5,Hangup Thanks in advance for help, Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR Timing out
Did what you suggested but with the same results plus the following error messages: [attendant] ;Main welcome message exten = s,1,Wait(2) exten = s,2,DigitTimeout,5 exten = s,3,AbsoluteTimeout(25) exten = s,4,Background(welcome_n2p1) ;exten = s,4,ResponseTimeout,25 ;exten = s,5,Hangup exten = s,5,Background(silence/10) exten = s,6,Background(silence/10) exten = s,7,Goto(s,5) Jan 24 13:10:51 WARNING[-1233134672]: file.c:473 ast_openstream: File silence/10 does not exist in any format Jan 24 13:10:51 WARNING[-1233134672]: file.c:761 ast_streamfile: Unable to open silence/10 (format ULAW): No such file or directory Kurt On Mon, 24 Jan 2005 12:53:11 -0500, Roger Gulbranson [EMAIL PROTECTED] wrote: On Mon, 2005-01-24 at 12:36 -0500, kurt x wrote: . Once the .gsm file is finished playing you can not select any of the menu items. The .gsm file is roughly 15 to 17 seconds long.If you make a selection before the .gsm file finishing playing you can select any of the menu items. OK, then following my previous advice, alter your s extension to: exten = s,1,Wait(2) exten = s,2,DigitTimeout(5) exten = s,3,AbsoluteTimeout(25) exten = s,4,Background(welcome_n2p1) exten = s,5,Background(silence/10) exten = s,6,Background(silence/10) exten = s,7,GoTo(s,5) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail.conf pin protection
Is there any way to encrypt the PIN numbers in voicemail.conf. I looked at the Wiki page for voicemail.conf but it did not mention anything about that topic. I am not using MySQL or any other thrid party database. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP debugs
Other then the standard sip debug is there any other sip debug bugs like for errors, events, etc. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Accessing Voice mail
Brain,
I did what you suggested but instead of going to VoiceMailMain it
starts the begining of
my recorded message each time I press the * key.
[vmail]
exten = a,1,Voicemail(u${ext})
exten = a,2,Hangup
Kurt
On Wed, 19 Jan 2005 11:48:18 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
If you put the following in your Dialplan, pressing * should break you
out of voicemail and call VoiceMailMain
exten = a,1,VoicemailMain,EXTEN
exten = a,2,Hangup
On Wed, 19 Jan 2005 11:33:23 -0500, kurt x [EMAIL PROTECTED] wrote:
I want to know if there is way to break out of the voicemail message.
for example:
On my Noterl PBX when you dial you number from any where
you get your recorded voice mail message, but during the message I
press 81 and break out of that message. It then
prompts me for my PIN thus allowing me to access my message
without using the auto attendant.
Is this possible with Comedian?
The below page did help.
http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMailMain
Kurt
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[Asterisk-Users] Accessing Voice mail
I want to know if there is way to break out of the voicemail message. for example: On my Noterl PBX when you dial you number from any where you get your recorded voice mail message, but during the message I press 81 and break out of that message. It then prompts me for my PIN thus allowing me to access my message without using the auto attendant. Is this possible with Comedian? The below page did help. http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMailMain Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound Dial via SIP
What I am trying to do is the following: A call is sent to the * box
via a SIP invite. The * box answers via an IVR menu system with
Enter the extension you want to dial so I enter in my 5 digit
extension and get the below message.
Jan 18 10:10:03 WARNING[-1380238416]: channel.c:1860 ast_request: No
channel type registered for 'SIP)'
Jan 18 10:10:03 NOTICE[-1380238416]: app_dial.c:696 dial_exec: Unable
to create channel of type 'SIP)'
Jan 18 10:10:05 WARNING[-1115923536]: chan_sip.c:673 retrans_pkt:
Maximum retries exceeded on call
[EMAIL PROTECTED]
for seqno 1 (Non-critical Response)
My extension.conf outbound dial peer:
[outbound]
exten = _124XX,1,Dial(SIP)/${EXTEN:[EMAIL PROTECTED])
exten = _124XX,2,Playback(invalid)
exten = _124XX,3,Hangup
My sip.conf
[outbound]
type=peer
host=192.168.1.1
What the * needs to do is receive the call via SIP and then send it
out dialed extension via SIP to an another IP PBX. SO the * does not
need to register to a server just blindly send a SIP invite to the ip
address in the SIP.CONF file: 192.168.1.1
Any help would be appricated
Kurt
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Re: [Asterisk-Users] Outbound Dial via SIP
That was the ticket. The Extra ) was the problem.
Thanks Sean.
Kurt
On Tue, 18 Jan 2005 08:13:31 -0800, Sean Kennedy [EMAIL PROTECTED] wrote:
kurt x wrote:
What I am trying to do is the following: A call is sent to the * box
via a SIP invite. The * box answers via an IVR menu system with
Enter the extension you want to dial so I enter in my 5 digit
extension and get the below message.
Jan 18 10:10:03 WARNING[-1380238416]: channel.c:1860 ast_request: No
channel type registered for 'SIP)'
Jan 18 10:10:03 NOTICE[-1380238416]: app_dial.c:696 dial_exec: Unable
to create channel of type 'SIP)'
Jan 18 10:10:05 WARNING[-1115923536]: chan_sip.c:673 retrans_pkt:
Maximum retries exceeded on call
[EMAIL PROTECTED]
for seqno 1 (Non-critical Response)
My extension.conf outbound dial peer:
[outbound]
exten = _124XX,1,Dial(SIP)/${EXTEN:[EMAIL PROTECTED])
exten = _124XX,2,Playback(invalid)
exten = _124XX,3,Hangup
My sip.conf
[outbound]
type=peer
host=192.168.1.1
What the * needs to do is receive the call via SIP and then send it
out dialed extension via SIP to an another IP PBX. SO the * does not
need to register to a server just blindly send a SIP invite to the ip
address in the SIP.CONF file: 192.168.1.1
Any help would be appricated
Kurt
exten = _124XX,1,Dial(SIP)/${EXTEN:[EMAIL PROTECTED])
This is your problem. It should be
exten = _124XX,1,Dial(SIP/${EXTEN}:[EMAIL PROTECTED])
There might be further syntax errors, this is only off the top of my head,
but the most glaring error that I could see was the ) after SIP.
Sean
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[Asterisk-Users] Directory() Command
I am trying to use the Directory() but am having difficulty using it. According to Wiki page that I found you need to pass it your voicemail.conf context. My vm-context is [local]. So when I setup the cmd (Directory(local)) I can search on the three letters of the last name find that user. But once I press one to except the name and dial the extension I get the following message form the * CLI. Jan 17 15:22:07 WARNING[-1285669968]: app_directory.c:182 play_mailbox_owner: Can't find extension '' in context 'local'. Did you pass the wrong context to Directory? Reading the above error message I see that I need to pass it my outbound context. So I setup the command to look as follows: Directory(local outbound). I reload * and try again but this time it does not even pick up the name I search for. I used the same name in the first example. Any ideas on where I want wrong would be greatly appreciated. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Cisco Unity and Asterisk
Question: What is your reasoning for using Cisco Voice Mail instead of Asterisk's voice mail. IMHO it would make more sense to keep everything on Asterisk. Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] supposable timing problem with TE100P
--On Thursday, November 04, 2004 04:41:53 PM +0100 Peter Svensson [EMAIL PROTECTED] wrote: On Thu, 4 Nov 2004, Kurt Bauer wrote: Is your timing source set correctly? If you are connecting to the pstn the pstn connection should be the primary timing source. connection is to a Ericsson MD110 wich is set as network, * is set as CPE. Have you set the span as the timing source? (second number in the span line in zaptel.conf). It was 0, I changed it to 1, which seems to work better, but I have to check the settings of the other side with our telephony guys What does the missed interrupts counter in cat cat /proc/zaptel/1 say? cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear (In use) 3 WCT1/0/3 Clear (In use) 4 WCT1/0/4 Clear (In use) Is there no IRQ miss counter for the E100P card? ??? I don't know. Thanks, Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] supposable timing problem with TE100P
Hi list, every now and then I get the following message in my * logs: chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 As this is only a notice and voice worked quite well, despite the messages, I didn't bother. _But_, I wanted to try spandsp/fax (latest version) lately and whenever I send a fax to * the pages get chopped, which according to Steve Underwood points at a timing problem, either in * or Hardware. I don't know if the messages about FCS errors have smth. to do with the fax problems, but hope that someone out there has a clue ;-)) BTW, I see a lot of the following messages too: !! Unknown IE 49 (cs5, Unknown Information Element) !! Unknown IE 50 (cs5, Unknown Information Element) If any further information is needed to narrow the problem down please let me know. Thanks a lot in advance, best regards, Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] supposable timing problem with TE100P
--On Thursday, November 04, 2004 03:19:56 PM +0100 Peter Svensson [EMAIL PROTECTED] wrote: On Thu, 4 Nov 2004, Kurt Bauer wrote: Hi list, every now and then I get the following message in my * logs: chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 As this is only a notice and voice worked quite well, despite the messages, I didn't bother. _But_, I wanted to try spandsp/fax (latest version) lately and whenever I send a fax to * the pages get chopped, which according to Steve Underwood points at a timing problem, either in * or Hardware. Is your timing source set correctly? If you are connecting to the pstn the pstn connection should be the primary timing source. connection is to a Ericsson MD110 wich is set as network, * is set as CPE. What does the missed interrupts counter in cat cat /proc/zaptel/1 say? cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear (In use) 3 WCT1/0/3 Clear (In use) 4 WCT1/0/4 Clear (In use) 5 WCT1/0/5 Clear (In use) 6 WCT1/0/6 Clear (In use) 7 WCT1/0/7 Clear (In use) 8 WCT1/0/8 Clear (In use) 9 WCT1/0/9 Clear (In use) 10 WCT1/0/10 Clear (In use) 11 WCT1/0/11 Clear (In use) 12 WCT1/0/12 Clear (In use) 13 WCT1/0/13 Clear (In use) 14 WCT1/0/14 Clear (In use) 15 WCT1/0/15 Clear (In use) 16 WCT1/0/16 HDLCFCS (In use) 17 WCT1/0/17 Clear (In use) 18 WCT1/0/18 Clear (In use) 19 WCT1/0/19 Clear (In use) 20 WCT1/0/20 Clear (In use) 21 WCT1/0/21 Clear (In use) 22 WCT1/0/22 Clear (In use) 23 WCT1/0/23 Clear (In use) 24 WCT1/0/24 Clear (In use) 25 WCT1/0/25 Clear (In use) 26 WCT1/0/26 Clear (In use) 27 WCT1/0/27 Clear (In use) 28 WCT1/0/28 Clear (In use) 29 WCT1/0/29 Clear (In use) 30 WCT1/0/30 Clear (In use) 31 WCT1/0/31 Clear (In use) br, Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: DTMF tones from CCME phone
You need to either download 12.3(11)T or 12.3(10)LD. Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Cisco to * problem
See if you have the below configure under your dial peers or voice service voip. If you do, then issue this command no signaling forward unconditional signaling forward unconditional Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP authentication problem
Hi, I have the following setup: E100P SER * - PBX This works just fine, except when there are users on both boxes (ie. SER and asterisk), whose usernames are the same, although the realm is different. An example: user '[EMAIL PROTECTED]' wants to call some extension in the PBX, but as user '[EMAIL PROTECTED]' exists too, * tries to authenticate the user, which it shouldn't do, at least I guess so. Shouldn't asterisk differentiate between the realms ie. [EMAIL PROTECTED] != [EMAIL PROTECTED] ? Find attached, the relevant part of the logged sip communication and the sip.conf. If you have any hints, please let me know. Thanks in advance, best regards, Kurt example sip.log Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Max-Forwards: 10 Record-Route: sip:[EMAIL PROTECTED];ftag=000cce3a7be800087fd8099f-62cc5396;lr=on Via: SIP/2.0/UDP 83.136.32.160;branch=z9hG4bK1cba.998c27f1.0 Via: SIP/2.0/UDP 131.130.220.101:5060;branch=z9hG4bK03521c86 From: Kurt Bauer sip:[EMAIL PROTECTED];tag=000cce3a7be800087fd8099f-62cc5396 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Mon, 06 Sep 2004 10:01:57 GMT CSeq: 102 INVITE User-Agent: CSCO/7 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Content-Type: application/sdp Content-Length: 253 v=0 o=Cisco-SIPUA 23148 13380 IN IP4 131.130.220.101 s=SIP Call c=IN IP4 131.130.220.101 t=0 0 m=audio 30596 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 15 headers, 11 lines Using latest request as basis request Sending to 83.136.32.160 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 131.130.220.101:30596 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x10f(G723|GSM|ULAW|ALAW|G729A), peer - audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0x10c(ULAW|ALAW|G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x0(EMPTY) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 83.136.32.160;branch=z9hG4bK1cba.998c27f1.0 Via: SIP/2.0/UDP 131.130.220.101:5060;branch=z9hG4bK03521c86 From: Kurt Bauer sip:[EMAIL PROTECTED];tag=000cce3a7be800087fd8099f-62cc5396 To: sip:[EMAIL PROTECTED];tag=as6191c2dd Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=troubadix.univie.ac.at, nonce=5276f268 Content-Length: 0 to 83.136.32.160:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user 'kb' troubadix*CLI Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 83.136.32.160;branch=z9hG4bK1cba.998c27f1.0 From: Kurt Bauer sip:[EMAIL PROTECTED];tag=000cce3a7be800087fd8099f-62cc5396 Call-ID: [EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as6191c2dd CSeq: 102 ACK User-Agent: Sip EXpress router(0.8.12-tcp_nonb-tls (i386/linux)) Content-Length: 0 8 headers, 0 lines /example sip.log --note the SIP/2.0 407 Proxy Authentication Required sip.conf ; ; SIP Configuration for Asterisk ; [general] port=5060 bindaddr=0.0.0.0 realm=troubadix.univie.ac.at disallow=all allow=ulaw allow=alaw allow=g729 allow=g723.1 allow=gsm [at43_in] type=peer host=sip.at43.at context=at43 insecure=very deny=0.0.0.0/0.0.0.0 permit=83.136.32.160/255.255.255.255 /sip.conf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Going to voicemail instead of queue if no agent is logged in ?
Hi,
I did this the following way:
-) define a global variable - AGENTS_AVAIL=0
-) when agent logs in increment -
SetGlobalVar(AGENTS_AVAIL=$[${AGENTS_AVAIL} + 1]);
-) when agent logs off decrement -
SetGlobalVar(AGENTS_AVAIL=$[${AGENTS_AVAIL} - 1]);
-) when queue is called evaluate and goto label -
gotoif,$[${AGENTS_AVAIL}]?${Q}:${NO_Q)
Hope that helps and if there is an easier way of doing this please show me
how.
br,
Kurt
--On Tuesday, August 31, 2004 09:57:29 PM +0200 Robert Rozman
[EMAIL PROTECTED] wrote:
Hi,
I'd like to implement scenario to send user to operator's queue by default
(if doesn't dial any extension) but only if there is operator agent
logged, so user could get response. If not I'd like to send it to
voicemail...
Any quick advice ?
Thanks in advance,
Robert.
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[Asterisk-Users] GUI VoiceMail directory question:
I would like to know where could I change the directory that the web interface looks to when trying to play wave file. The current location is as follows: /var/spool/asterisk/vm/1234/INBOX/msg.wav When I use addmailbox command it places it under /var/spool/asterisk/voicemail/default/1234/. How can I change the WEB base interface to point to the voicemail directory? I do not want to use a symbolic link to do this. Thanks, Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] strange problem PBX-Asterisk
Thanks for the hints, 'overlapdial=yes' did the trick. br, kurt --On Tuesday, August 24, 2004 10:08:08 PM +0200 Peter Svensson [EMAIL PROTECTED] wrote: On Tue, 24 Aug 2004, Christian Victor wrote: maybe I oversee somth. very obvious, but I'm a little puzzled about the following 'error': When I make a call from the PBX to * I get number not available, but on debug I see, that asterisk is searching just for the first digit in the extension, which of course doesn't exist, eg: I seems that you PBX uses Overlap Dial and transmits the extensien digit by digit and Asterisk expects the extension to be en bloc. So when it receives anything from the PBX (wich is in this case the first digit) Asterisk thinks that this is the whole block of extension. Don't know how to fix it though. ;-) This in case you are on a PRI: see overlapdial at http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf It needs to be set on pri links where ovarlap dialing is used, even incoming towards asterisk. Without more information on the connections between the systems and the configuration it is hard to figure out what is wrong. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX and backup Circuits
I am interested to know how one would calculate the amount of PSTN connection needed for backup on an Asterisk PBX that is being setup to receive its DIDs via a VoIP provide. To sum up what I am implementing: I am porting my DIDs to a VoIP provide so I will need a back up plan in place if the Data network fails. In addition, 911 will always be going out the PSTN so I know I need at least one POTs circuit. Calls inbound and outbound will always routed through the data network. Thanks, Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX and backup Circuits
I am interested to know how one would calculate the amount of PSTN connection needed for backup on an Asterisk PBX that is being setup to receive its DIDs via a VoIP provide. To sum up what I am implementing: I am porting my DIDs to a VoIP provide so I will need a back up plan in place if the Data network fails. In addition, 911 will always be going out the PSTN so I know I need at least one POTs circuit. Calls inbound and outbound will always routed through the data network. Thanks, Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] strange problem PBX-Asterisk
Hi, maybe I oversee somth. very obvious, but I'm a little puzzled about the following 'error': When I make a call from the PBX to * I get number not available, but on debug I see, that asterisk is searching just for the first digit in the extension, which of course doesn't exist, eg: I dial 77 (for conn to *) and 12345 (valid extension) on the console I see: -- Extension '1' in context 'sip-local' from '+ 14070' does not exist. Rejecting call on channel 0/1, span 1 Everything worked fine before an update on Friday, but I haven't changed any config files then. I 'downgraded' to 1.0RC2 today, but then same problem. If any of you has any hints, please let me know. Thanks a lot, br, Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel Problem after Upgrade
Hi, after upgrading to latest CVS (20.08) I have a problem with the connection to PSTN: When I try to make a call from PSTN to * I hear the number not available' sound and the following warnings on * console: Aug 20 12:09:55 WARNING[180235]: chan_zap.c:6762 zt_pri_error: PRI: !! Unknown IE 1329 (len = 3) Aug 20 12:09:55 WARNING[180235]: chan_zap.c:6762 zt_pri_error: PRI: !! Unknown IE 1330 (len = 3) The other way round still works although I also get a warning: Aug 20 12:10:39 WARNING[180235]: chan_zap.c:6762 zt_pri_error: PRI: !! Unknown IE 1330 (len = 3) I changed no config files, so that couldn't be the problem, or has anything changed since April (that was the CVS version with which it worked just fine). Thanks, best regards, Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Possible Asterisk Notify Bug
I noticed when my Cisco device sends a SUBSCRIBE message to Asterisk for voice mail subscription. The Asterisk server will send the wrong call ID back. Thus, the Cisco sends a 481 back to the Asterisk. I believe the below section in RFC 3265 is relevant: 3.3.4 NOTIFY requests are matched to such SUBSCRIBE requests if they contain the same Call-ID, a To header tag parameter which matches the From header tag parameter of the SUBSCRIBE, and the same Event header field. Rules for comparisons of the Event headers are described in section 7.2.1. If a matching NOTIFY request contains a Subscription-State of active or pending, it creates a new subscription and a new dialog (unless they have already been created by a matching response, as described above). Below is a portion of the trap I obtained from the Asterisk Server. A complete trap can be found at http://www.pasewaldt.com/notify.html Sip read: SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bKFC2 From: 2486 sip:[EMAIL PROTECTED];tag=6C30-149 To: sip:[EMAIL PROTECTED] Date: Call-ID: 2C1ED0F6-2BDE11D6-80048294-A080CC2F CSeq: 101 SUBSCRIBE Timestamp: 1089723760 Contact: sip:[EMAIL PROTECTED]:5060 Event: message-summary Expires: 600 Accept: application/simple-message-summary Content-Length: 0 13 headers, 0 lines ^Dasterick*CLI Using latest SUBSCRIBE request as basis request Sending to 192.168.0.1 : 5060 (non-NAT) Looking for 2486 in voice-mail Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK19c0b453 From: asterisk sip:[EMAIL PROTECTED];tag=as288443e6 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 37 Messages-Waiting: yes Voicemail: 7/0 (no NAT) to 192.168.0 Kurt __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
