[asterisk-users] WebRTC and JsSIP
Hi ! My name is Gerald and I am working with WEBRTC and JsSIP.I configure my Asterisk 11.7.0 to work wit WEBRTC.Using a JsSIP (http://tryit.jssip.net/), the SIP extension can connect at the Asterisk, but when we try to make a call they send a 488 response and finish it.here is the part of the SIP DEBUG<--- SIP read from WS:177.64.122.237:49217 --->BYE sip:500@187.122.82.197:0;transport=ws SIP/2.0Via: SIP/2.0/WS e8ilhkrhlup2.invalid;branch=z9hG4bK7306188Max-Forwards: 69To: ;tag=as52a1a298From: "G" ;tag=ue84kn6rkuCall-ID: u5hkiispkvn9g841oedeCSeq: 9338 BYEReason: SIP ;cause=488; text="Not Acceptable Here"Supported: path, outbound, gruuUser-Agent: JsSIP 0.3.7Content-Length: 0Some one can help me with this problem?Thanks Gerald -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Number of Calls
Hi, I am new in voip, how many calls can one asterisk box handle with 30 % of trans-coded calls and system configuration as 8GB RAM X3430 Xeon Processor, 2.4GHz, 8M Cache, Turbo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ayrv2by jg4yjbf3r
w1z7g0t, 2ck5wt7y6. http://au6vpf8so.blog.com/1d/ fyooxwq sl5pk8 8unmhkev, tudcx e5zxhd. 62ce7jtt 9ygow7phv8b. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] wctdm24xx IRQ missing
Hello all, I just instaled a tdm2400 Digium card on my asterisk box. When it boots, I can see some error messages in "dmesg". wctdm24xxp :21:08.0: Missed interrupt. Increasing latency to 8 ms in order to compensate. wctdm24xxp :21:08.0: Missed interrupt. Increasing latency to 10 ms in order to compensate. wctdm24xxp :21:08.0: Missed interrupt. Increasing latency to 11 ms in order to compensate. wctdm24xxp :21:08.0: Missed interrupt. Increasing latency to 12 ms in order to compensate. wctdm24xxp :21:08.0: Missed interrupt. Increasing latency to 13 ms in order to compensate. I try to change the IRQ of the card, using the "setpci" command, without success. I am using dadhi 2.2.1 and asterisk 1.4.24 Has anyone the same problem? How could I change to fix this errors? Any clue will be wellcomed. Voipcrazy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DIALSTATUS on CANCEL
Hello,
We have a strange situation (asterisk 1.6.2.14), where we get a result for
DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.
This is the (relevant) test dialplan:
[incoming-private]
exten => _X., n, Dial(SIP/1001,30)
exten => _X., n, NoOp(${DIALSTATUS})
exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1)
[incoming-status]
exten => s-CANCEL,1, NoOp()
exten => s-CANCEL,n, Return()
exten => s-NOANSWER,1, NoOp()
exten => s-NOANSWER,n, Return()
exten => s-BUSY,1, NoOp()
exten => s-BUSY,n, Return()
This is what we get on a BUSY call:
---
-- Executing [1...@incoming-private:3] Dial("SIP/Proxy-002b",
"SIP/1001,50") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 1001
-- Got SIP response 486 "Busy Here" back from 10.0.0.1
-- SIP/1001-002c is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [1...@incoming-private:4] NoOp("SIP/Proxy-002b",
"BUSY") in new stack
-- Executing [1...@incoming-private:5] Gosub("SIP/Proxy-002b",
"incoming-status,s-BUSY,1") in new stack
This is what we get on a NO ANSWER call:
---
-- Executing [1...@incoming-private:3] Dial("SIP/Proxy-002f",
"SIP/1001,30") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 1001
-- SIP/1001-0030 is ringing
-- Nobody picked up in 3 ms
-- Executing [1...@incoming-private:4] NoOp("SIP/Proxy-002f",
"NOANSWER") in new stack
-- Executing [1...@incoming-private:5] Gosub("SIP/Proxy-002f",
"incoming-status,s-NOANSWER,1") in new stack
This is what we get on a CANCEL call:
-
-- Executing [1...@incoming-private:3] Dial("SIP/Proxy-0031",
"SIP/1001,30") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 1001
-- SIP/1001-0032 is ringing
== Spawn extension (incoming-private, , 3) exited non-zero on
'SIP/Proxy-0031'
There's no event indicating that a DIALSTATUS is generated and the call
simply doesn't go to the next step in the dialplan. Unless I'm missing
something, it seems to me that it might be a bug.
I would be happy to get feedback from other users of the DIALSTATUS value
(or Digium), especially in the CANCEL scenario.
Thank you,
Michael
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[asterisk-users] Asterisk Log viewer
Hello, I want to analyze the asterisk logs files, looking for all kind of errors, ¿Anyboby knows any asterisk logs analyzer? Thanks all, Voipcrazy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI SUBSCRIBE Settings
Hello list members, We're trying to get MWI notifications on our ATA device and we set it to send SUBSCRIBE messages to Asterisk, but it gets UNAUTHORIZED messages, despite the fact that we set the following lines in its settings in sip.conf: subscribemwi=yes mailbox...@from-extensions We need help in understanding how this works and what we are doing wrong. This is the SIP debug we get: <--- SIP read from UDP:10.0.0.4:5090 ---> SUBSCRIBE sip:2...@10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5090;rport;branch=z9hG4bK3663664e35 From: ;tag=6d8c6ac6 To: Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 Contact: CSeq: 1 SUBSCRIBE Max-Forwards: 70 Expires: 60 Accept: application/simple-message-summary Event: message-summary User-Agent: CM5K-TA2S (810170) Content-Length: 0 <-> --- (13 headers 0 lines) --- Creating new subscription Sending to 10.0.0.4 : 5090 (no NAT) list_route: hop: Found peer '21' for '21' from 10.0.0.4:5090 <--- Transmitting (no NAT) to 10.0.0.4:5090 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.4:5090;branch=z9hG4bK3663664e35;received=10.0.0.4;rport=5090 From: ;tag=6d8c6ac6 To: ;tag=as25bc6135 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 CSeq: 1 SUBSCRIBE Server: S-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="ePBX", nonce="26c43866" Content-Length: 0 <> Scheduling destruction of SIP dialog '055f7edd4081e1ec0f176e0a4b395...@10.0.0.4' in 6400 ms (Method: SUBSCRIBE) <--- SIP read from UDP:10.0.0.4:5090 ---> SUBSCRIBE sip:2...@10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5090;rport;branch=z9hG4bK3663664e35 From: ;tag=6d8c6ac6 To: Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 Contact: CSeq: 1 SUBSCRIBE Max-Forwards: 70 Expires: 60 Accept: application/simple-message-summary Event: message-summary User-Agent: CM5K-TA2S (810170) Content-Length: 0 <-> --- (13 headers 0 lines) --- Ignoring this SUBSCRIBE request Found peer '21' for '21' from 10.0.0.4:5090 <--- Transmitting (no NAT) to 10.0.0.4:5090 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.4:5090;branch=z9hG4bK3663664e35;received=10.0.0.4;rport=5090 From: ;tag=6d8c6ac6 To: ;tag=as25bc6135 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 CSeq: 1 SUBSCRIBE Server: S-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="ePBX", nonce="26c43866" Content-Length: 0 <> Scheduling destruction of SIP dialog '055f7edd4081e1ec0f176e0a4b395...@10.0.0.4' in 6400 ms (Method: SUBSCRIBE) <--- SIP read from UDP:10.0.0.4:5090 ---> SUBSCRIBE sip:2...@10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5090;rport;branch=z9hG4bK3663664e35 From: ;tag=6d8c6ac6 To: Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 Contact: CSeq: 1 SUBSCRIBE Max-Forwards: 70 Expires: 60 Accept: application/simple-message-summary Event: message-summary User-Agent: CM5K-TA2S (810170) Content-Length: 0 <-> --- (13 headers 0 lines) --- Ignoring this SUBSCRIBE request Found peer '21' for '21' from 10.0.0.4:5090 <--- Transmitting (no NAT) to 10.0.0.4:5090 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.4:5090;branch=z9hG4bK3663664e35;received=10.0.0.4;rport=5090 From: ;tag=6d8c6ac6 To: ;tag=as25bc6135 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 CSeq: 1 SUBSCRIBE Server: S-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="ePBX", nonce="26c43866" Content-Length: 0 <--- SIP read from UDP:10.0.0.4:5090 ---> SUBSCRIBE sip:2...@10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5090;rport;branch=z9hG4bK3663664e35 From: ;tag=6d8c6ac6 To: Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 Contact: CSeq: 1 SUBSCRIBE Max-Forwards: 70 Expires: 60 Accept: application/simple-message-summary Event: message-summary User-Agent: CM5K-TA2S (810170) Content-Length: 0 <-> --- (13 headers 0 lines) --- Ignoring this SUBSCRIBE request Found peer '21' for '21' from 10.0.0.4:5090 <--- Transmitting (no NAT) to 10.0.0.4:5090 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.4:5090;branch=z9hG4bK3663664e35;received=10.0.0.4;rport=5090 From: ;tag=6d8c6ac6 To: ;tag=as25bc6135 Call-ID: 055f7edd4081e1ec0f176e0a4b395...@10.0.0.4 CSeq: 1 SUBSCRIBE Server: S-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="ePBX", nonce="26c43866" Content-Length: 0 <> Scheduling destruction of SIP dialog '055f7edd4081e1ec0f176e0a4b395...@10.0.0.4' in 6400 ms (Method: SUBSCRIBE) Thanks, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introduc
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Thanks Kevin,
We managed to get the ReceiveFAX going, while making some minor changes
to the code, like, for example, using the ${UNIQUEID} for the file name.
Regards,
Michael
Original Message
Subject: Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not
acceptable here)
From: Kevin P. Fleming
To: asterisk-users@lists.digium.com
Date: Thursday, 21 October, 2010 16:13:02
> On 10/20/2010 09:35 AM, VoIP Question wrote:
>> Thank you Kevin,
>>
>> We'll upgrade our server to 1.6.2.12 and try again.
>>
>> Another question: Is there (expect for the admin guide that we didn't
>> succeed to understand the example in) an example somewhere for
>> ReceiveFax full extensions.conf diaplan? We would like to allocate one
>> of the extensions that our SIP provider gives us to a fax storage server
>> or later to email.
>
> The ReceiveFAX example in the Fax For Asterisk administrator's guide is
> very straightforward and easy to follow... if you don't understand it,
> then you'll need to spend some time learning how the Asterisk dialplan
> works. I would highly recommend reading the O'Reilly Asterisk book
> (which you can read online for free)... while it is based on Asterisk
> 1.4, the dialplan concepts documented in it have not changed much in
> Asterisk 1.6, and gaining that basic understanding will go a long way
> towards helping you be able to resolve these issues on your own.
>
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Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
We're learning all the time and made some significant progress and some very nice calls scenarios, but specifically with this issue, is there anything we can do to solve the interop problem with this end-point? Thanks. Original Message Subject: Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here) From: Kevin P. Fleming To: asterisk-users@lists.digium.com Date: Thursday, 21 October, 2010 16:11:00 > On 10/20/2010 11:35 AM, VoIP Question wrote: >> On Wed, Oct 20, 2010 at 4:25 PM, Kevin P. Fleming> <mailto:kpflem...@digium.com>> wrote: >> >> >> This was fixed in Asterisk 1.6.2.12 and later releases, so if you were >> running the current version, you wouldn't have experienced this specific >> problem. This was listed in the ChangeLog for 1.6.2.12, but >> unfortunately the commit message the developer wrote did not explain why >> the change was made or what problem it was addressing, so you wouldn't >> have noticed it. >> >> In any case, upgrading to 1.6.2.12 or later will cure this problem. >> >> I upgraded to 1.6.2.13 and now we get this error (with a specific >> destination, to which we occasionally need to send faxes): >> >> WARNING[857]: udptl.c:1087 ast_udptl_write: (SIP/XXX): UDPTL >> asked to send 50 bytes of IFP when far end only prepared to accept 30 >> bytes; data loss will occur.You may need to override the >> T38FaxMaxDatagram value for this endpoint in the channel driver >> configuration. >> >> >> How can we fix it, without risking incompatibility with other >> end-points? What's a "channel driver configuration" and where is it? > > It appears that you need to spend some time learning the basics of > Asterisk. In this case, the channel driver is chan_sip, since the > channel involved is a SIP channel, and the 'channel driver > configuration' is the sip.conf file. It is unfortunate that you have > chosen to tackle a very complex task (T.38 interoperability is fraught > with problems due to widely varying implementations) as your first > experience with Asterisk... there's a lot you'll need to learn to be > able to diagnose and troubleshoot problems. Asterisk alone is not 'point > and click', and adding T.38 to the mix makes things more complicated. > -- _ -- 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] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Hello again, If I set a peer to use G.711 only, they try to process a sent fax in G.711, but Asterisk doesn't like it: WARNING[4903]: res_fax.c:1709 sendfax_t38_init: Audio FAX not allowed on channel 'SIP/Main-000a' and T.38 negotiation failed; aborting. What can I do to enable it? Thanks, Michael -- _ -- 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] 2 step dialing
On Wed, Oct 20, 2010 at 6:11 PM, Danny Nicholas wrote:
> Should be (_X.,15,Dial(Local/1/${ccde...@outgoing,50)
>
> Or
>
> _X.,15,Dial(Local/${ccde...@outgoing,50)
>
We used the {} (without the extra 1) and it works perfectly. Thank you
>
>
> ${ccdest} returns value of ccdest to dialplan
>
> Local/${ccdest} selects local channel ccdest
>
> You might want local/1 with ccdest value.
>
We didn't understand the need for the /1, but it's not relevant, as it works
now
>
>
> Also you might want to look into the DISA command
>
Nice to know, but we built it (with CLI list and passwords) in less than 15
lines...
Thanks again for your help
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Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Wed, Oct 20, 2010 at 4:25 PM, Kevin P. Fleming wrote: > > This was fixed in Asterisk 1.6.2.12 and later releases, so if you were > running the current version, you wouldn't have experienced this specific > problem. This was listed in the ChangeLog for 1.6.2.12, but > unfortunately the commit message the developer wrote did not explain why > the change was made or what problem it was addressing, so you wouldn't > have noticed it. > > In any case, upgrading to 1.6.2.12 or later will cure this problem. > > I upgraded to 1.6.2.13 and now we get this error (with a specific destination, to which we occasionally need to send faxes): WARNING[857]: udptl.c:1087 ast_udptl_write: (SIP/XXX): UDPTL asked to send 50 bytes of IFP when far end only prepared to accept 30 bytes; data loss will occur.You may need to override the T38FaxMaxDatagram value for this endpoint in the channel driver configuration. How can we fix it, without risking incompatibility with other end-points? What's a "channel driver configuration" and where is it? Thank you. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2 step dialing
Hello all, We're trying to build a small IVR application to allow callers to use the Asterisk for outgoing calls in a 2 steps dialing mode. The context for outgoing calls is called "outgoing" (we have there an LCR and routing mechanism we want to use, depending on the destination). This is what we did, but it doesn't work: exten => _X., 13, Read(ccdest,vm-enter-num-to-call,,,2) exten => _X., 14, NoOp($ccdest) exten => _X., 15, Dial(Local/$ccd...@outgoing,50) The error we get is: chan_local.c:538 local_call: No such extension/context $ccd...@outgoingwhile calling Local channel -- Couldn't call $ccd...@outgoing We know there's a syntax problem in line 15, but not sure how to fix it. Thank you for your help. Michael -- _ -- 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] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Thank you Kevin, We'll upgrade our server to 1.6.2.12 and try again. Another question: Is there (expect for the admin guide that we didn't succeed to understand the example in) an example somewhere for ReceiveFax full extensions.conf diaplan? We would like to allocate one of the extensions that our SIP provider gives us to a fax storage server or later to email. Michael On Wed, Oct 20, 2010 at 4:25 PM, Kevin P. Fleming wrote: > > You have a 'Local' channel in between SendFAX and the SIP channel to > your other endpoint. In Asterisk 1.6.2.11, chan_local was not properly > aware of T.38 negotiation, so it ends up acting as a sort of 'firewall' > between the endpoints. > > This was fixed in Asterisk 1.6.2.12 and later releases, so if you were > running the current version, you wouldn't have experienced this specific > problem. This was listed in the ChangeLog for 1.6.2.12, but > unfortunately the commit message the developer wrote did not explain why > the change was made or what problem it was addressing, so you wouldn't > have noticed it. > > In any case, upgrading to 1.6.2.12 or later will cure this problem. > > -- _ -- 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] Using Calls Rejection Reasons
Thank you guys for the quick answer. To complicate it a bit, is there a way to extract the rejection reason (in this case 486) without using the DIALSTATUS and to use it? It can be useful for rejections that are not handled by DIALSTATUS. On Wed, Oct 20, 2010 at 4:05 PM, bakko wrote: > Yes, > > look at DIALSTATUS variable that Asterisk set when use DIAL Application. > > Regards > > - Bakko > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using Calls Rejection Reasons
Hello all, We would like to "inform" the caller of the reason for a failed call. For example, when we get a "486 Busy Here", the system accepts it and in the CLI we see "Everyone is busy/congested at this time". Can we use this data to play an announcement to the caller? Thank you in advance for your help. Michael -- _ -- 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] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
To: ;tag=gK028217ef
Contact:
Call-ID: 2d965b0926e0134e0b211f882cbd2...@yyy.yyy.yyy.yyy
CSeq: 102 ACK
User-Agent: PBX
Content-Length: 0
---
sip*CLI> -- SIP/main-002a answered Local/12345...@outgoing-2c36;2
sip*CLI>> Channel Local/12345...@outgoing-2c36;1 was answered.
-- Executing [...@outboundfax:1] Set("Local/12345...@outgoing-2c36;1",
"FAXOPT(filename)=/tmp/1585867851.tif") in new stack
[Oct 19 18:41:17] WARNING[31185]: res_fax.c:2454 acf_faxopt_write: channel
'Local/12345...@outgoing-2c36;1' set FAXOPT(filename) to
'/tmp/1585867851.tif' is unhandled!
-- Executing [...@outboundfax:2] Set("Local/12345...@outgoing-2c36;1",
"FAXOPT(ecm)=yes") in new stack
sip*CLI> -- Executing [...@outboundfax:3]
Set("Local/12345...@outgoing-2c36;1", "FAXOPT(headerinfo)=Smartel") in new
stack
-- Executing [...@outboundfax:4] Set("Local/12345...@outgoing-2c36;1",
"FAXOPT(localstationid)=+972-72-278-0008") in new stack
-- Executing [...@outboundfax:5] Set("Local/12345...@outgoing-2c36;1",
"FAXOPT(maxrate)=14400") in new stack
-- Executing [...@outboundfax:6] Set("Local/12345...@outgoing-2c36;1",
"FAXOPT(minrate)=2400") in new stack
-- Executing [...@outboundfax:7] SendFAX("Local/12345...@outgoing-2c36;1",
"/tmp/1585867851.tif,d") in new stack
-- Channel 'Local/12345...@outgoing-2c36;1' sending FAX:
--/tmp/1585867851.tif
sip*CLI> -- Channel 'Local/12345...@outgoing-2c36;1' FAX session '6'
started
sip*CLI> -- FAX handle 0: [ 000.000152 ], STAT_EVT_STRT_TX st:
IDLE rt: IDLENSTX
-- FAX handle 0: [ 000.000214 ], STAT_EVT_TX_HW_RDY st: WT_TX_HW_RDY
rt: TRDYNHTY
-- FAX handle 0: [ 000.000240 ], P30EVN_SEND_STARTED
sip*CLI> == Spawn extension (outgoing, 12345678, 1) exited non-zero on
'Local/12345...@outgoing-2c36;2'
sip*CLI>> Channel 'Local/12345...@outgoing-2c36;1' fax session '6',
[ 000.075767 ], channel sent 2 frames (40 ms) of silence.
sip*CLI>> Channel 'Local/12345...@outgoing-2c36;1' fax session '6',
[ 000.588736 ], stack sent 28 frames (560 ms) of energy.
sip*CLI>> Channel 'Local/12345...@outgoing-2c36;1' fax session '6',
[ 003.566428 ], stack sent 149 frames (2980 ms) of silence.
sip*CLI>> Channel 'Local/12345...@outgoing-2c36;1' fax session '6',
[ 004.089420 ], stack sent 26 frames (520 ms) of energy.
sip*CLI>
<--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --->
INVITE sip:98765...@10.0.0.3:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK02B020504a6f7a14f9a
From: ;tag=gK028217ef
To: "Fax" ;tag=as28606a47
Call-ID: 2d965b0926e0134e0b211f882cbd2...@yyy.yyy.yyy.yyy
CSeq: 8530 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay, multipart/mixed
Contact:
Supported: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Length: 306
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 28160 32051 IN IP4 xxx.xxx.xxx.xx8
s=SIP Media Capabilities
c=IN IP4 xxx.xxx.xxx.xx7
t=0 0
m=image 6256 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:262
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv
<->
--- (16 headers 13 lines) ---
Sending to xxx.xxx.xxx.xx8 : 5060 (no NAT)
Got T.38 offer in SDP in dialog
2d965b0926e0134e0b211f882cbd2...@yyy.yyy.yyy.yyy
Capabilities: us - 0x102 (gsm|g729), peer - audio=0x0 (nothing)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
<--- Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
xxx.xxx.xxx.xx8:5060;branch=z9hG4bK02B020504a6f7a14f9a;received=xxx.xxx.xxx.xx8
From: ;tag=gK028217ef
To: "Fax" ;tag=as28606a47
Call-ID: 2d965b0926e0134e0b211f882cbd2...@yyy.yyy.yyy.yyy
CSeq: 8530 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact:
Content-Length: 0
<>
sip*CLI>
<--- Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
xxx.xxx.xxx.xx8:5060;branch=z9hG4bK02B020504a6f7a14f9a;received=xxx.xxx.xxx.xx8
From: ;tag=gK028217ef
To: "Fax" ;tag=as28606a47
Call-ID: 2d965b0926e0134e0b211f882cbd2...@yyy.yyy.yyy.yyy
CSeq: 8530 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OP
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Digium claims that their FFA is the best and most compatible solution and they give one channel for free, but do not provide support for those that do not buy more channels, but why buy more channels if the free/test one doesn't work? I know they read (and sometimes respond) to this list, so I don't understand why they don't clarify this issue. I spent a few hours on Google and saw many similar posts, but no actual valuable answer. Weird... On Tue, Oct 19, 2010 at 6:08 PM, Danny Nicholas wrote: > From what I have read over the last few months, you should invest in > Motrin before trying T.38 faxing with or without FFA – it can (possibly) be > done, but it has beaten some folks into the ground trying it. > > > > Could be a codec issue. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
We don't have an ATA and fax machine. The whole point (as I specified in the header and initial message) is the attempt to use "Fax for Asterisk" to send the message. As I showed in the logs, the remote carrier sends a proper T.38 reINVITE, but our Asterisk doesn't accept, despite the fact that this provider is defined in sip.conf with both canreinvite and t38pt_udptl enabled, so the only question is (as far as we understand) is why in this scenario, the T.38 is rejected. Here are the logs (sip debug is open) again, since we get the reINVITE: <--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---> INVITE sip:1234...@10.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8 From: ;tag=gK0d817deb To: "Fax" ;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: Supported: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Content-Length: 303 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 218 7126 IN IP4 xxx.xxx.xxx.xx8 s=SIP Media Capabilities c=IN IP4 xxx.xxx.xxx.xx7 t=0 0 m=image 6202 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:262 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy a=sendrecv <-> --- (16 headers 13 lines) --- Sending to xxx.xxx.xxx.xx8 : 5060 (no NAT) Got T.38 offer in SDP in dialog 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy Capabilities: us - 0x102 (gsm|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. <--- Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8 From: ;tag=gK0d817deb To: "Fax" ;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Server: Smartel-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <> <--- Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8 From: ;tag=gK0d817deb To: "Fax" ;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Server: Smartel-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Thanks. Michael On Tue, Oct 19, 2010 at 5:40 PM, David Backeberg wrote: > On Tue, Oct 19, 2010 at 11:21 AM, VoIP Question > wrote: > > It's set to yes for this peer. > > > > also t38pt_udptl is set to yes. > > > > :( > > You don't say anything about what you're trying to send / receive against. > > Here's how you should troubleshoot: > > * start with a 'real fax machine' if you have one, on an analog line > if you have one. If you can't receive / send with that against your > target, blame your target. > * move to audio-pass through fax on asterisk. No T.38. If that works. > * add in T.38 > > You will learn things in that process and be able to tell at what > layer your troubles are happening. > > It could be coincidental that things give up during the reinvite. It > could actually be giving up for noise on the line, packet drops, etc. > > At the very least, start recording the call. You'll at least be able > to hear up to the re-invite. > > Definitely record the audio passthrough attempt and listen back to it. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
It's set to yes for this peer. also t38pt_udptl is set to yes. :( On Tue, Oct 19, 2010 at 5:12 PM, David Backeberg wrote: > On Tue, Oct 19, 2010 at 10:36 AM, VoIP Question > wrote: > > Hello, > > > > I'm trying to send a tif file, using Fax for Asterisk and the call is > > executed, but when I get the reINVITE with T.38 data, the local server > > doesn't recognize that we have this capability and sends a 488 message. > > http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite > > take a look at your canreinvite option. > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Hello, I'm trying to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. These are the logs: <--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---> INVITE sip:1234...@10.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8 From: ;tag=gK0d817deb To: "Fax" ;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: Supported: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Content-Length: 303 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 218 7126 IN IP4 xxx.xxx.xxx.xx8 s=SIP Media Capabilities c=IN IP4 xxx.xxx.xxx.xx7 t=0 0 m=image 6202 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:262 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy a=sendrecv <-> --- (16 headers 13 lines) --- Sending to xxx.xxx.xxx.xx8 : 5060 (no NAT) Got T.38 offer in SDP in dialog 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy Capabilities: us - 0x102 (gsm|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. <--- Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8 From: ;tag=gK0d817deb To: "Fax" ;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Server: Smartel-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <> <--- Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8 From: ;tag=gK0d817deb To: "Fax" ;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Server: Smartel-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Please help. Thank you. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Checking SIP Headers existence and content
Hello, I would like to verify if a specific SIP header exists, and if yes, extract the partial content from another header. 1. Is there a way to verify if a specific header exists? 2. How do I extract data that is between the first : and the following @? Specifically, The data looks like and I would like to get only the 1234567890 I tried to use the CUT() command, but without success. Thank you. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Switchboad like application
Hello all, Anybody could point me any clue about an Open Source or licensed switchboard for my users? ARI or FOP is not enought for my users. Thanks in advance. VoipCrazy -- _ -- 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] Qwest PRIs
Ok got it up and running. In the case for Qwest with NFAS they reserve what they call "Interface ID 1" for the circuit with the backup d channel. In our case we only have two circuits with a single d channel. The real key was realizing the logical span number in the spanmap translated into "interface ID" so here are the spanmaps that worked for us: [trunkgroups] trunkgroup => 1,24 spanmap => 1,1,0 spanmap => 2,1,2 group=1 switchtype=dms100 echocancel=yes signalling=pri_cpe channel =>1-23,25-48 Notice the switchtype. While they told me that their switchtype was "NI2" (National ISDN 2) they did say they were using a dms100, so I would always ask your carrier what switch they have. Anyway thanks for your help all. On Mon, Jun 14, 2010 at 4:47 PM, Steve Edwards wrote: > On Mon, 14 Jun 2010, C F wrote: > > > One more thing, read the comments here: > > > http://www.voip-info.org/wiki/index.php?page_id=573&tk=2ff846f8169b7694aed5&comments_page=1 > > Don't forget to have a beer ready :P > > Now that's really funny. > > I read along with this and was thinking this was exactly my experience > with some Qwest PRIs a couple of years ago. > > Then I noticed -- it was me :) > > I guess I had too many beers. > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Qwest PRIs
Ya I'm passed that part now. I have dahdi properly loading the card, and both links are green. Asterisk recognizes the channels, but still shows the span as down. On Sun, Jun 13, 2010 at 6:35 PM, C F wrote: > Not sure what version you are running but I'm still running 1.2x in > 1.2 you can't bring up PRI outside asterisk, since the PRI (I'm > assuming layer 2+) part loads with Asterisk. > > On Sat, Jun 12, 2010 at 10:51 AM, Voip Asterisk > wrote: > > Ya i'm not even to the asterisk part yet. I'm still trying to get dahdi > to > > bring up the PRIs without alarms. > > > > On Sat, Jun 12, 2010 at 4:58 AM, Doug Lytle wrote: > >> > >> Voip Asterisk wrote: > >> > Hi, > >> > > >> > I'm trying to bring up two PRIs from qwest with asterisk and dahdi. > >> > I'm using an OpenVox D410E and the drivers are loaded. My > >> > system.conf looks like this: > >> > > >> > >> Google turned up this: > >> > >> http://www.voip-info.org/wiki/view/NFAS > >> > >> Doug > >> > >> -- > >> Ben Franklin quote: > >> > >> "Those who would give up Essential Liberty to purchase a little > Temporary > >> Safety, deserve neither Liberty nor Safety." > >> > >> > >> -- > >> _ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> New to Asterisk? Join us for a live introductory webinar every Thurs: > >> http://www.asterisk.org/hello > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Qwest PRIs
Ya 99% sure that isn't it since they were just pulled working off an AS5300 On Sun, Jun 13, 2010 at 4:27 AM, Doug Lytle wrote: > Voip Asterisk wrote: > > > > Status: Provisioned, Down, Active > > > > specifically the "Down" part. > > > > > In my experience that usually means the provider hasn't brought up the > PRI. Granted, I've never used anything beyond a single PRI. > > Doug > > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- 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] Qwest PRIs
Ok making a little progress, but still can't get them completely up: pri show span 1 Primary D-channel: 24 Status: Provisioned, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 Logical Channel Mapping: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 Overlap Recv: No cat /proc/dahdi/1; cat /proc/dahdi/2 Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) B8ZS/ESF CRC4 error count: 1293 E-bit error count: 2774 1 TE4/0/1/1 Clear (In use) 2 TE4/0/1/2 Clear (In use) 3 TE4/0/1/3 Clear (In use) 4 TE4/0/1/4 Clear (In use) 5 TE4/0/1/5 Clear (In use) 6 TE4/0/1/6 Clear (In use) 7 TE4/0/1/7 Clear (In use) 8 TE4/0/1/8 Clear (In use) 9 TE4/0/1/9 Clear (In use) 10 TE4/0/1/10 Clear (In use) 11 TE4/0/1/11 Clear (In use) 12 TE4/0/1/12 Clear (In use) 13 TE4/0/1/13 Clear (In use) 14 TE4/0/1/14 Clear (In use) 15 TE4/0/1/15 Clear (In use) 16 TE4/0/1/16 Clear (In use) 17 TE4/0/1/17 Clear (In use) 18 TE4/0/1/18 Clear (In use) 19 TE4/0/1/19 Clear (In use) 20 TE4/0/1/20 Clear (In use) 21 TE4/0/1/21 Clear (In use) 22 TE4/0/1/22 Clear (In use) 23 TE4/0/1/23 Clear (In use) 24 TE4/0/1/24 HDLCFCS (In use) Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" B8ZS/ESF 25 TE4/0/2/1 Clear (In use) 26 TE4/0/2/2 Clear (In use) 27 TE4/0/2/3 Clear (In use) 28 TE4/0/2/4 Clear (In use) 29 TE4/0/2/5 Clear (In use) 30 TE4/0/2/6 Clear (In use) 31 TE4/0/2/7 Clear (In use) 32 TE4/0/2/8 Clear (In use) 33 TE4/0/2/9 Clear (In use) 34 TE4/0/2/10 Clear (In use) 35 TE4/0/2/11 Clear (In use) 36 TE4/0/2/12 Clear (In use) 37 TE4/0/2/13 Clear (In use) 38 TE4/0/2/14 Clear (In use) 39 TE4/0/2/15 Clear (In use) 40 TE4/0/2/16 Clear (In use) 41 TE4/0/2/17 Clear (In use) 42 TE4/0/2/18 Clear (In use) 43 TE4/0/2/19 Clear (In use) 44 TE4/0/2/20 Clear (In use) 45 TE4/0/2/21 Clear (In use) 46 TE4/0/2/22 Clear (In use) 47 TE4/0/2/23 Clear (In use) 48 TE4/0/2/24 Clear (In use) cat system.conf | grep -v "#" span=1,1,1,esf,b8zs bchan=1-23 dchan=24 span=2,2,1,esf,b8zs bchan=25-48 loadzone= us defaultzone = us cat chan_dahdi.conf [trunkgroups] trunkgroup => 1,24 spanmap => 1,1,1 spanmap => 2,1,2 [channels] context=incoming switchtype=national facilityenable=no signalling=pri_cpe usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 channel => 1-23 group=2 channel => 25-48 If anyone could let me know what I should be doing next. I'm sure my issue is: Status: Provisioned, Down, Active specifically the "Down" part. Thanks On Sat, Jun 12, 2010 at 9:04 AM, Voip Asterisk wrote: > BTW these were just up and running on a Cisco AS5300, so we know they work > and Qwest has them turned up and configured correctly. The only settings > that were needed on the cisco was the coding and framing which is how dahdi > is configured now. Anyone have any idea how to mimic default cisco settings > on dahdi/asterisk? > > The line card used in the cisco was the standard 4 port T1 PRI card. > > Thanks > > On Sat, Jun 12, 2010 at 7:51 AM, Voip Asterisk wrote: > >> Ya i'm not even to the asterisk part yet. I'm still trying to get dahdi >> to bring up the PRIs without alarms. >> >> >> On Sat, Jun 12, 2010 at 4:58 AM, Doug Lytle wrote: >> >>> Voip Asterisk wrote: >>> > Hi, >>> > >>> > I'm trying to bring up two PRIs from qwest with asterisk and dahdi. >>> > I'm using an OpenVox D410E and the drivers are loaded. My >>> > system.conf looks like this: >>> > >>> >>> Google turned up this: >>> >>> http://www.voip-info.org/wiki/view/NFAS >>> >>> Doug >>> >>> -- >>> Ben Franklin quote: >>> >>> "Those who would give up Essential Liberty to purchase a little Temporary >>> Safety, deserve neither Liberty nor Safety." >>> >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Aster
Re: [asterisk-users] Qwest PRIs
BTW these were just up and running on a Cisco AS5300, so we know they work and Qwest has them turned up and configured correctly. The only settings that were needed on the cisco was the coding and framing which is how dahdi is configured now. Anyone have any idea how to mimic default cisco settings on dahdi/asterisk? The line card used in the cisco was the standard 4 port T1 PRI card. Thanks On Sat, Jun 12, 2010 at 7:51 AM, Voip Asterisk wrote: > Ya i'm not even to the asterisk part yet. I'm still trying to get dahdi to > bring up the PRIs without alarms. > > > On Sat, Jun 12, 2010 at 4:58 AM, Doug Lytle wrote: > >> Voip Asterisk wrote: >> > Hi, >> > >> > I'm trying to bring up two PRIs from qwest with asterisk and dahdi. >> > I'm using an OpenVox D410E and the drivers are loaded. My >> > system.conf looks like this: >> > >> >> Google turned up this: >> >> http://www.voip-info.org/wiki/view/NFAS >> >> Doug >> >> -- >> Ben Franklin quote: >> >> "Those who would give up Essential Liberty to purchase a little Temporary >> Safety, deserve neither Liberty nor Safety." >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- _ -- 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] Qwest PRIs
Ya i'm not even to the asterisk part yet. I'm still trying to get dahdi to bring up the PRIs without alarms. On Sat, Jun 12, 2010 at 4:58 AM, Doug Lytle wrote: > Voip Asterisk wrote: > > Hi, > > > > I'm trying to bring up two PRIs from qwest with asterisk and dahdi. > > I'm using an OpenVox D410E and the drivers are loaded. My > > system.conf looks like this: > > > > Google turned up this: > > http://www.voip-info.org/wiki/view/NFAS > > Doug > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Qwest PRIs
Hi, I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm using an OpenVox D410E and the drivers are loaded. My system.conf looks like this: # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED span=1,2,0,esf,b8zs bchan=1-24 # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF RED span=2,1,0,esf,b8zs bchan=25-47 dchan=48 These are suppose to be configured with NFAS (Non-Facilities Associated Signaling) so 1 d channels manages all the b channels in both spans. I can't seem to get it out of alarm RED, asterisk running or not doesn't make a difference. cat /proc/dahdi/1*;cat /proc/dahdi/2* Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED 1 TE4/0/1/1 Clear RED 2 TE4/0/1/2 Clear RED 3 TE4/0/1/3 Clear RED 4 TE4/0/1/4 Clear RED 5 TE4/0/1/5 Clear RED 6 TE4/0/1/6 Clear RED 7 TE4/0/1/7 Clear RED 8 TE4/0/1/8 Clear RED 9 TE4/0/1/9 Clear RED 10 TE4/0/1/10 Clear RED 11 TE4/0/1/11 Clear RED 12 TE4/0/1/12 Clear RED 13 TE4/0/1/13 Clear RED 14 TE4/0/1/14 Clear RED 15 TE4/0/1/15 Clear RED 16 TE4/0/1/16 Clear RED 17 TE4/0/1/17 Clear RED 18 TE4/0/1/18 Clear RED 19 TE4/0/1/19 Clear RED 20 TE4/0/1/20 Clear RED 21 TE4/0/1/21 Clear RED 22 TE4/0/1/22 Clear RED 23 TE4/0/1/23 Clear RED 24 TE4/0/1/24 Clear RED Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF RED 25 TE4/0/2/1 Clear RED 26 TE4/0/2/2 Clear RED 27 TE4/0/2/3 Clear RED 28 TE4/0/2/4 Clear RED 29 TE4/0/2/5 Clear RED 30 TE4/0/2/6 Clear RED 31 TE4/0/2/7 Clear RED 32 TE4/0/2/8 Clear RED 33 TE4/0/2/9 Clear RED 34 TE4/0/2/10 Clear RED 35 TE4/0/2/11 Clear RED 36 TE4/0/2/12 Clear RED 37 TE4/0/2/13 Clear RED 38 TE4/0/2/14 Clear RED 39 TE4/0/2/15 Clear RED 40 TE4/0/2/16 Clear RED 41 TE4/0/2/17 Clear RED 42 TE4/0/2/18 Clear RED 43 TE4/0/2/19 Clear RED 44 TE4/0/2/20 Clear RED 45 TE4/0/2/21 Clear RED 46 TE4/0/2/22 Clear RED 47 TE4/0/2/23 Clear RED 48 TE4/0/2/24 HDLCFCS RED Any ideas why these are alarmed? Also what settings need to be set in chan_dahdi.conf for asterisk to bring these up? Thanks Miles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP ports
Hello, I need to limit the RTP ports used by an asterisk in a client, Actualy the range defined is from 1 to 2 udp ports. If I only have 10 local sip extension ¿how many ports/range should I set up in /etc/asterisk/rtp.conf? Which is the way to calculate the rtp ports needed on an instalation? Thanks in advance, Voipcrazy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom Provisioning
Hello all, I've to deploy about 200 snom320 phones on a instalation. Do you know any knid of tool to help me with this amount of phones? I'm thinking in a provisioning tool which I use for setting up the phones. Any clue would be welcomed. Thanks. Voip-Crazy -- _ -- 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] ReceiveFAX G.711 + Realtime
> Now, referring to the error above, I see (in voip-info.org) that
> t38passthrough is an R/O variable and not an R/W, but in any case, I got
> 0 as a result, so it should have been OK, and it's not, as ReceiveFAX
> still sends a T.38 reINVITE. If I can't modify it, what should I do?
For the testing, I set the peer's t38pt_udptl to no and on the
originating gateway, left only aLaw enabled. If/when I set it back to
yes, it sent the reINVITE, so I don't have a solution for that yet.
>
> Also, since the rtpdest includes also the port, how do I check in the
> GotoIf if the value "contains" that IP and not equal to it (which it
> can't be)? It seems that this will always return 0:
> $[${CHANNEL(rtpdest)}=xx.xxx.xxx.xxx]
I changed the GotoIf command to
$[${CHANNEL(rtpdest):0:14}=xx.xxx.xxx.xxx]?5:T38,1 as I don't care about
the length of the other originating IP addresses (the specific one I
check is always 14 chars long), and now it stays in in aLaw context, but
now I get this error:
app_fax.c:292 fax_generator_generate: Only generating 240 samples, where
320 requested
I see that in app_fax.c MAX_SAMPLES is set to 240 and it doesn't seem to
accept config file values. I reduced the packet length from 40ms (320
samples) to 20ms (160 samples) for G.711 codecs and it solved this
problem. Does this mean that Asterisk supports maximum 30ms packets in
G.711 Fax?
Thanks.
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Re: [asterisk-users] ReceiveFAX G.711 + Realtime
> I have no idea where you got the idea that such a thing is possible...
> it's not. sip.conf settings for SIP endpoints are not channel variables,
> and cannot be modified from the dialplan unless the CHANNEL() dialplan
> function has been specifically extended to support them.
I was actually HOPING that it was possible, while guessing it probably
isn't ;-), at least not like I did it.
> If you don't
> want T.38 support for a SIP endpoint, don't configure it that way in
> sip.conf or in your Realtime SIP peers table.
I want this "endpoint" to support T.38, but what I actually want is to
check the initial INVITE's SDP and based on the IP address of the media,
make a dialplan rule to decide whether to use G.711 or to switch to
T.38. So, I found the CHANNEL() variables "rtpdest" and "t38passthrough".
This is the dialplan real-time script I ran:
"id","context","exten","priority","app","appdata"
80,"fax","aLaw",1,"NoOp","${CHANNEL(t38passthrough)}"
81,"fax","aLaw",2,"NoOp","${CHANNEL(rtpdest)}"
82,"fax","aLaw",3,"Set","CHANNEL(t38passthrough)=0"
83,"fax","aLaw",4,"GotoIf","$[${CHANNEL(rtpdest)}=xx.xxx.xxx.xxx]?5:T38,1"
76,"fax","aLaw",5,"Set","SIP_CODEC=aLaw"
77,"fax","aLaw",6,"Answer",""
78,"fax","aLaw",7,"ReceiveFAX","${fax_filepath}/${UNIQUEID}.tif"
79,"fax","aLaw",8,"Hangup",""
The result was:
-- Executing NoOp("SIP/Proxy-0005", "0")
-- Executing NoOp("SIP/Proxy-0005", "xx.xxx.xxx.xxx:60100")
-- Executing Set("SIP/Proxy-0005", "CHANNEL(t38passthrough)=0")
[2009-12-29 16:20:19.772] WARNING[5888]: func_channel.c:161
func_channel_write: Unknown or unavailable item requested: 't38passthrough'
-- Executing GotoIf("SIP/Proxy-0005", "0?5:T38,1")
-- Goto (fax,T38,1)
Now, referring to the error above, I see (in voip-info.org) that
t38passthrough is an R/O variable and not an R/W, but in any case, I got
0 as a result, so it should have been OK, and it's not, as ReceiveFAX
still sends a T.38 reINVITE. If I can't modify it, what should I do?
Also, since the rtpdest includes also the port, how do I check in the
GotoIf if the value "contains" that IP and not equal to it (which it
can't be)? It seems that this will always return 0:
$[${CHANNEL(rtpdest)}=xx.xxx.xxx.xxx]
Thanks.
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[asterisk-users] ReceiveFAX G.711 + Realtime
Hello,
We're trying to receive G.711 (aLaw) faxes on the asterisk and convert
them to tif. With T.38, we have several issues, so we are trying to use
G.711, since the gateway is located in the same LAN, so there's no
bandwidth/packet-lose issue.
We also use on the same Asterisk Real-Time process for the extensions.conf
My question:
Is the following syntax for disabling T.38 support correct?
vm*CLI> -- Executing Set("SIP/Proxy-", "t38pt_udptl=no")
vm*CLI> -- Executing Set("SIP/Proxy-", "SIP_CODEC=aLaw")
vm*CLI> -- Executing Answer("SIP/Proxy-", "")
The aLaw Set command is taken into consideration, because the SDP of the
OK that follows these lines includes only codec 8, but when the
ReceiveFAX command is executed, Asterisk immediately sends a T.38 reINVITE:
vm*CLI> -- Executing ReceiveFAX("SIP/Proxy-",
"/var/spool/asterisk/fax/3/1261993891.0.tif")
I must be doing something wrong, but I'm not sure what :-(
Your help would be highly appreciated.
Thanks,
Andreas
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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9
> Cyprus VoIP wrote: > >> This is the reINVITE SDP received from the SIP Proxy: >> --- >> Content-Type: application/sdp >> Content-Length: 353 >> >> v=0 >> o=root 30427 30428 IN IP4 194.98.xxx.xxx >> s=session >> c=IN IP4 194.98.xxx.xxx >> t=0 0 >> m=image 17548 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:72 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> --- > > This is probably originating from a Cisco gateway. Cisco gateways > generate T.38 SDPs that do not conform to the T.38 recommendation in one > very obvious (and painful) way: they tell us that they can only accept > 72 byte packets (T38FaxMaxDatagram), when in fact they can accept > packets much larger than that. When you notice that they are also > requesting that we use t38UDPRedundancy for error correction, that means > that the maximum IFP (single FAX protocol packet) we can include in a > UDPTL datagram is around 30 bytes, since we'd need to have room for two > of them and a bit of overhead. 30 bytes is a ridiculously small limit > for IFPs, and does not allow successful FAXing at any possible bit rate > (except for 2400 bits per second using 10 millisecond IFPs, but no FAX > stack would do that). > > There is code in Asterisk already to deal with this problem, however... > see below. > >> pbx*CLI> Got UDPTL packet from 192.168.2.11:9100 (type 0, seq 0, len 86) >> [Dec 15 12:38:05] WARNING[5262]: udptl.c:997 ast_udptl_write: UDPTL >> asked to send 77 bytes of IFP when far end only prepared to accept 12 >> bytes; data loss may occur. You may need to override the >> T38FaxMaxDatagram value for this endpoint in the channel driver >> configuration. > > Have you followed these instructions? The message is fairly clear in > describing the problem, and the description of how and why this is > needed is spelled out in the sip.conf.sample file in the configs > directory of the source tree. > > Setting a lower limit for the max datagram value used when communicating > with this peer (and others like it that generate incorrect > T38FaxMaxDatagram values) will resolve this problem. > Hi, Yes. I saw the message and the required addition in the sip.conf. The problem is that if I set it to 72, other terminating gateways that support 400 or more would also be limited to 72. What doesn't make sense is Asterisk's "commission". Why doesn't it simply pass/use whatever it gets without cutting the values? It looks like it's a bug. Alternatively, I could (had I known how ;-)) set this ATA not to relay the RTP via Asterisk, in which case maybe Asterisk would leave the values unchanged. When I use another port on this ATA from the same location to the same destination without passing through the Asterisk, the faxes go through. How should I define this peer in sip.conf so that Asterisk wouldn't relay the RTP for it? Thanks, Andreas ___ -- 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] Fax throughput - Asterisk 1.6.1.9
Hello, We upgraded the Asterisk to 1.6.1.11. Now, there's no RTP reINVITE, but the datagram handling of Asterisk is strange. Basically, it "takes a commission" from both ends, and ends up overflowing: Reminder, we're dealing in this example with a passthrough, where we have an ATA device connected to Asterisk in the same LAN, the Asterisk is registered to a remote SIP Proxy server and behind it, a Fax server. This is the reINVITE SDP received from the SIP Proxy: --- Content-Type: application/sdp Content-Length: 353 v=0 o=root 30427 30428 IN IP4 194.98.xxx.xxx s=session c=IN IP4 194.98.xxx.xxx t=0 0 m=image 17548 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy --- Asterisk sends this reINVITE SDP to the ATA device (notice that the datagram was reduced by 2): --- Content-Type: application/sdp Content-Length: 269 v=0 o=root 31812 318120001 IN IP4 192.168.2.10 s=Asterisk PBX 1.6.1.11 c=IN IP4 192.168.2.10 t=0 0 m=image 4427 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:70 a=T38FaxUdpEC:t38UDPRedundancy --- Then, it gets OK SDP from the ATA with the same settings it suggested: --- Content-Type: application/sdp Content-Length: 275 v=0 o=101 01 02 IN IP4 192.168.2.11 s=A conversation c=IN IP4 192.168.2.11 t=0 0 m=image 9100 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:70 a=T38FaxUdpEC:t38UDPRedundancy a=sendrecv --- But, when it sends the OK SDP to the remote end, it lowers the datagram again: --- Content-Type: application/sdp Content-Length: 275 v=0 o=root 936220937 936220938 IN IP4 xxx.xxx.xxx.xxx s=Asterisk PBX 1.6.1.11 c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=image 4650 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:65 a=T38FaxUdpEC:t38UDPRedundancy --- Then, when the ATA device sends T.38 packets, it freaks outs: --- pbx*CLI> Got UDPTL packet from 192.168.2.11:9100 (type 0, seq 0, len 86) [Dec 15 12:38:05] WARNING[5262]: udptl.c:997 ast_udptl_write: UDPTL asked to send 77 bytes of IFP when far end only prepared to accept 12 bytes; data loss may occur. You may need to override the T38FaxMaxDatagram value for this endpoint in the channel driver configuration. [Dec 15 12:38:05] ERROR[5262]: udptl.c:291 encode_open_type: Buffer overflow detected (77 + 3 > 72) [Dec 15 12:38:05] NOTICE[5262]: udptl.c:1010 ast_udptl_write: UDPTL Transmission error to 194.98.xxx.xxx:17548: Message too long Sent UDPTL packet to 194.98.xxx.xxx:17548 (type 0, seq 35, len -1) pbx*CLI> Got UDPTL packet from 192.168.2.11:9100 (type 0, seq 0, len 162) [Dec 15 12:38:05] WARNING[5262]: udptl.c:997 ast_udptl_write: UDPTL asked to send 77 bytes of IFP when far end only prepared to accept 12 bytes; data loss may occur. You may need to override the T38FaxMaxDatagram value for this endpoint in the channel driver configuration. [Dec 15 12:38:05] ERROR[5262]: udptl.c:291 encode_open_type: Buffer overflow detected (77 + 3 > 72) [Dec 15 12:38:05] NOTICE[5262]: udptl.c:1010 ast_udptl_write: UDPTL Transmission error to 194.98.xxx.xxx:17548: Message too long --- Thank you for your kind assistance and support. Regards, Andreas Original Message Subject: Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9 From: Cyprus VoIP To: Asterisk Users Mailing List - Non-Commercial Discussion Date: Friday, 04 December, 2009 18:21:59 >> It's probably because you are using 1.6.1.9; that release (and older) >> had a 'feature' that allowed automatic switching back to audio from T.38 >> if one of the endpoints sent an audio packet. It turns out that wasn't a >> good idea, and it's been removed... but in later versions. You'll have >> to update to the latest release to get that fixed. >> > > Will do. Thanks for the explanation. ___ -- 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] ATA FXO
Joseph, You may want to try RPA-2E1S1O from www.broad-tel.com from China. It provides real FXO port that registers with Asterisk. David On Sat, Dec 12, 2009 at 1:37 AM, Joseph wrote: > I'm looking for a reliable ATA FXO/FXS adapter. > > Linksys 3102 - a lot of echo problem + two of them died within a year (not > reliable) > Sangoma USBFXO - problem installing drive in Gentoo. > > I've tried two Chines units: AG-188N and YGW30B > none are of them have real FXO port that will register with Asterisk. > > Any other recommendations; (I don't like internal cards). > > -- > Joseph > > ___ > -- 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] Asterisk 1.6.1.11 Fax
>> We're trying to receive faxes on the Asterisk server, but for the time >> being T.38 negotiation fails. >> >> The SDP that the Asterisk reINVITE sends contains these lines: >> -- >> m=image 4968 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval >> a=T38FaxTranscodingMMR >> a=T38FaxTranscodingJBIG >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxDatagram:1400 >> a=T38FaxUdpEC:t38UDPRedundancy >> -- >> >> The MaxDatagram and MaxBitRate are definitely not what they should be, >> and maybe other parameters are also wrong. I would like to have 400 and >> 14400 respectively. > > There is no point in having a smaller T38FaxMaxDatagram value; > Asterisk's FAX applications can handle larger packets than 400 bytes, so > Asterisk is telling the sending endpoint that it can send them if it > wishes to do so. > > As far as the T38MaxBitRate, that is only a suggestion in the > negotiation, and rarely has any effect on the negotiation process or on > the resulting FAX transmission. > > As before, you've provided only a very small amount of information, not > enough to be able to help you determine what is wrong. Nothing in that > SDP offer indicate any problems of any kind. In addition, you haven't > indicated which FAX applications in Asterisk you are using (app_fax or > res_fax), which could also have an impact on the T.38 negotiation process. > > If you'd like people to be able to help you debug problems, you need to > provide enough information for them to do so; most of us are not > clairvoyant, telepathic or omniscient. In any situation where T.38 > negotiation is failing, that means a 'sip set debug on' log trace that > shows the entire T.38 negotiation transaction, ideally with 'core set > debug 10' and 'core set verbose 10' as well so we can see all the > actions that Asterisk and the FAX application took during the > negotiation process. > Hi Kevin, thank you for your response. Before posting my question, I analyzed the entire SIP negotiation and it was fine. The problem began in the T.38 negotiation itself, after the Asterisk's reINVITE. I don't have the old calls traces anymore, but I'll make new ones tomorrow and post them here. Regarding the MaxBitRate, is it possible to increase it to 14400 or is 9600 the max allowed? Regards, Andreas ___ -- 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] Asterisk 1.6.1.11 Fax
Hello, We're trying to receive faxes on the Asterisk server, but for the time being T.38 negotiation fails. The SDP that the Asterisk reINVITE sends contains these lines: -- m=image 4968 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy -- The MaxDatagram and MaxBitRate are definitely not what they should be, and maybe other parameters are also wrong. I would like to have 400 and 14400 respectively. That's the udptl.conf: -- ; ; UDPTL Configuration (UDPTL is one of the transports for T.38) ; [general] ; ; UDPTL start and UDPTL end configure start and end addresses ; udptlstart=4000 udptlend=4999 ; ; Whether to enable or disable UDP checksums on UDPTL traffic ; ;udptlchecksums=no ; ; The number of error correction entries in a UDPTL packet ; udptlfecentries = 3 ; ; The error correction type to be sent ; ;T38FaxUdpEC = t38UDPFEC T38FaxUdpEC = t38UDPRedundancy ; ; The maximum length of a UDPTL packet ; ;T38FaxMaxBuffer = 200 T38FaxMaxDatagram = 400 VoipFaxMaxRate = 5 ; The span over which parity is calculated for FEC in a UDPTL packet ; udptlfecspan = 3 ; ; Some VoIP providers will only accept an offer with an even-numbered ; UDPTL port. Set this option so that Asterisk will only attempt to use ; even-numbered ports when negotiating T.38. Default is no. use_even_ports = yes -- Can anyone help us identify the problem? Thanks, Andreas ___ -- 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] Realtime Database Tables
Thanks Fred, I'm actually there, but I was wondering if the tables there are up to date and if any changes took place. I see all kinds of comments about changes. Original Message Subject: Re: [asterisk-users] Realtime Database Tables From: Fred Posner To: Asterisk Users Mailing List - Non-Commercial Discussion Date: Thursday, 10 December, 2009 05:26:07 > On Dec 9, 2009, at 10:15 PM, Cyprus VoIP wrote: > >> Hello, >> >> We just installed a new 1.6.1.11 system + 1.6.1.2 addons and we would >> like to use the sip,extensions and voicemail in realtime mode. >> >> Where can we find the database tables structure for these versions? >> >> Thanks, >> >> Andreas >> > > This is the first place to go: > > http://www.voip-info.org/wiki/view/Asterisk+RealTime > > ___ > -- 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] Realtime Database Tables
Hello, We just installed a new 1.6.1.11 system + 1.6.1.2 addons and we would like to use the sip,extensions and voicemail in realtime mode. Where can we find the database tables structure for these versions? Thanks, Andreas ___ -- 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] Fax throughput - Asterisk 1.6.1.9
> It's probably because you are using 1.6.1.9; that release (and older) > had a 'feature' that allowed automatic switching back to audio from T.38 > if one of the endpoints sent an audio packet. It turns out that wasn't a > good idea, and it's been removed... but in later versions. You'll have > to update to the latest release to get that fixed. > Will do. Thanks for the explanation. ___ -- 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] Fax throughput - Asterisk 1.6.1.9
> Cyprus VoIP wrote: > >> So, I enabled the full logger, and the strange thing I see is this message: >> "Got T.38 Re-invite without audio. Keeping RTP active during T.38 session" >> >> It seems that this might be the reason Asterisk initiates a reINVITE >> with voice codecs, after connecting the 2 parties. > > Sorry, that's not the issue. That just means that chan_sip didn't > destroy the internal RTP structures used for the audio part of the call > when the call switched to T.38, which is only an optimization so we > don't have to recreate them if the call switches back. > Hi Kevin, Thank you for your support. If it's not related, why does Asterisk send again INVITE messages to both parties? How can this be prevented? I don't see more debug data prior to the new INVITE. 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] Fax throughput - Asterisk 1.6.1.9
> Set 'canreinvite=no' on all applicable peers? > I tried with yes and no. No difference. I'm almost certain it's related to the "Keeping RTP active during T.38 session" issue. ___ -- 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] Fax throughput - Asterisk 1.6.1.9
> Cyprus VoIP wrote: > >> Thank you for your answer. The 'internal extension' is indeed a T.38 >> capable device that works perfectly when connected directly to the >> Proxy/ITSP. >> >> As you said, the key to debugging/resolving this issue is the logger. I >> wasn't aware of this file. this is what I have there: >> ... >> ;debug => debug >> console => notice,warning,error >> ;console => notice,warning,error,debug >> messages => notice,warning,error >> ;full => notice,warning,error,debug,verbose >> ... >> >> Should I change the "console..." line or uncomment the ";full..." line? > > Either one is fine; using 'full' is actually a bit better, because the > color highlighting done on the console sometimes makes console captures > hard to read. > Hi, So, I enabled the full logger, and the strange thing I see is this message: "Got T.38 Re-invite without audio. Keeping RTP active during T.38 session" It seems that this might be the reason Asterisk initiates a reINVITE with voice codecs, after connecting the 2 parties. Is there a way to disable that action, or do we need to add T.38 somehow to the list of codecs? I followed the instructions on the default sip.conf to include the line "t38pt_udptl=yes,redundancy" in the general section and in each of the parties. 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] Fax throughput - Asterisk 1.6.1.9
>> We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the >> local extension and remote Proxy, but it still forces the call to go >> back to a voice call. > > Define 'internal extension'. Is this a T.38-capable device? If not, > Asterisk doesn't support TDM-to-T.38 FAX relay (yet). If it is, then the > path to resolving this problem is to collect a complete console log of > the failing call, including 'core set debug 10', 'core set verbose 10' > and 'sip set debug on' (and please ensure that all five logger levels > are enabled for the 'console' log channel in logger.conf). > Hi Kevin, Thank you for your answer. The 'internal extension' is indeed a T.38 capable device that works perfectly when connected directly to the Proxy/ITSP. As you said, the key to debugging/resolving this issue is the logger. I wasn't aware of this file. this is what I have there: ... ;debug => debug console => notice,warning,error ;console => notice,warning,error,debug messages => notice,warning,error ;full => notice,warning,error,debug,verbose ... Should I change the "console..." line or uncomment the ";full..." line? Regards, Andreas ___ -- 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] Fax throughput - Asterisk 1.6.1.9
Hello, We are trying to send faxes by T.38 protocol to a remote SIP proxy from a local extension. The local extension sends the INVITE, Asterisk sends the call to the Proxy the call is connected with a regular audio codec. After a few seconds the remote proxy sends an INVITE with UDPTL and the Asterisk sends it to the local extension and it's accepted, but (here the problem starts) just after sending the OK with the proper SDP to the remote Proxy, the Asterisk initiates a new INVITE to the local extension and remote Proxy, with the normal audio codecs again. We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the local extension and remote Proxy, but it still forces the call to go back to a voice call. Any idea why it happens and how to debug it? We set verbose and debug to 20, but no "internal" info is provided to get a clear understanding on Asterisk's "thoughts" during that process. Thank you in advance for your assistance, Andreas ___ -- 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] Asterisk 1.6.1.9 with FreePBX 2.5.2.1
>> The problem is that the online module update is not working for me >> (Cannot connect to online repository (mirror.freepbx.org). Online >> modules are not available.) and I couldn't find online a working >> solution :-( > > DNS/Gateway ok on server? > Yes. The problem is with the FreePBX modules. I forced the mirror file to include version 2.5, and I get a list, but when I try to install the modules, it says that the modules need FreePBX version 2.5.0alpha or rc1 or higher, but although 2.5.2 is indeed higher, it's rejected. I've given up on this software and will continue to edit my .conf files manually. what a waste of time :-( ___ -- 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] Asterisk 1.6.1.9 with FreePBX 2.5.2.1
>> I tried to install Asterisk + Asterisk addons + FreePBX (latest >> versions >> of all), but in the FreePBX screen, I don't have the option to set >> ring >> groups and IVRs >> >> Can anyone tell me what I'm doing wrong? > > You are not posting on the FreePBX forums? ;) > I figured "Asterisk-Users" would know ;) > > The solution however, is to install the modules using the module admin. > The problem is that the online module update is not working for me (Cannot connect to online repository (mirror.freepbx.org). Online modules are not available.) and I couldn't find online a working solution :-( > > Steve > ___ -- 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] Asterisk 1.6.1.9 with FreePBX 2.5.2.1
Hello, I tried to install Asterisk + Asterisk addons + FreePBX (latest versions of all), but in the FreePBX screen, I don't have the option to set ring groups and IVRs . Can anyone tell me what I'm doing wrong? Thanks, Andreas ___ -- 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] Music On Hold
Hello all, I adressed this issue a couple of weekes ago, but didn't find a solution yet. It seems that MOH is not initiated by Asterisk version 1.6.1.6 when receiving "sendonly INVITE", in order to put a call on hold. I configured features.conf with the following settings, and *9 initiates a proper HOLD with MOH, and so does *0 for blind transfer, but the a=sendonly doesn't: [featuremap] blindxfer => *0 atxfer => *9 Does anyone know if there's a parameter to set or if it's a bug in this version? Thanks. Original Message Subject: Re: [asterisk-users] Music On Hold From: Cyprus VoIP To: Asterisk Users Mailing List - Non-Commercial Discussion Date: Saturday, 03 October, 2009 09:28:20 > > > What does your musiconhold.conf look like? > > > > > [general] > > [default] > mode=files > directory=/var/lib/asterisk/moh > ___ -- 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] Billing applications
Hello all, I want to instal a Billing solution in the same asterisk's box. I have browse for ast2bill asterisk billing, astercc, and more, bu ti do not know which will be the best for me. The only things i need, are, - Postpaid and prepaid applications. - True CDR. Better that asterisk one, With suport for transfers - I do not need support for reseller - Billing for Voip, PSTN trunks I need a light app. I'm not searching a heavy app. with a lots of modules and applicacions. I need a ligth application for a soho and its needs. Any one are using a billing application which fits this needs? Any clue will be welcomed. Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Billing applications
Hello all, I want to instal a Billing solution in the same asterisk's box. I have browse for ast2bill asterisk billing, astercc, and more, bu ti do not know which will be the best for me. The only things i need, are, - Postpaid and prepaid applications. True CDR, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
> What does your musiconhold.conf look like? > [general] [default] mode=files directory=/var/lib/asterisk/moh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
>
> What is the output of "moh files show" CLI command ?
>
pbx*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/music_100
File: /var/lib/asterisk/moh/CHANGES-asterisk-moh-opsound-2
File: /var/lib/asterisk/moh/CREDITS-asterisk-moh-opsound-2
File: /var/lib/asterisk/moh/LICENSE-asterisk-moh-opsound-2
One more thing I tried is to add the m option in the dial command, to
check what happens, when the call is initially originated:
exten => ,n,Dial(SIP/21&SIP/22&SIP/23&SIP/24&SIP/25,,m(default))
This is what the console shows me:
-- Executing [1...@default:6] Dial("SIP/IN-PROXY-08563908",
"SIP/21&SIP/22&SIP/23&SIP/24&SIP/25,,m(default)") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 21
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 22
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 23
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 24
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 25
-- Started music on hold, class 'default', on SIP/IN-PROXY-08563908
-- SIP/23-b7c2d928 is ringing
-- SIP/24-b7a7b3a8 is ringing
-- SIP/22-b7c35cd8 is ringing
-- SIP/21-b7c3a508 is ringing
-- SIP/25-b7a83350 is ringing
-- Stopped music on hold on SIP/IN-PROXY-08563908
== Spawn extension (default, 99935709, 6) exited non-zero on
'SIP/IN-PROXY-08563908'
The music is played instead of the RBT, but during the conversation,
when put on hold, I only get silence and I don't get any reference in
the console to the fact that the call has been put on hold.
Thanks.
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Re: [asterisk-users] Music On Hold
Hi,
I deleted all the default files and put one that I know that works on
another Asterisk, but since then, I recompiled Asterisk and the default
files were added.
In order to test moh, I created a context for it:
[default]
exten => 888,1,Goto(moh,s,1)
[moh]
exten => s,1,Answer
exten => s,2,MusicOnHold()
When we dial 888, we hear the music and this appears in the console:
-- Executing [...@default:1] Goto("SIP/24-08650e80", "moh,s,1") in
new stack
-- Goto (moh,s,1)
-- Executing [...@moh:1] Answer("SIP/24-08650e80", "") in new stack
-- Executing [...@moh:2] MusicOnHold("SIP/24-08650e80", "") in new stack
-- Started music on hold, class 'default', on SIP/24-08650e80
-- Stopped music on hold on SIP/24-08650e80
== Spawn extension (moh, s, 2) exited non-zero on 'SIP/24-08650e80'
But, when I just put a call on hold, nothing is played and nothing
appears in the console.
I have no idea why this happens and what to do about it. Any suggestions?
Thanks.
Original Message
Subject: Re: [asterisk-users] Music On Hold
From: John A. Sullivan III
To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Wednesday, 30 September, 2009 15:27:28
> On Wed, 2009-09-30 at 14:57 +0300, Cyprus VoIP wrote:
>
>> > You see the wav files but do you see the files encoded for the codecs
>> you are using?
>> There's only one wav file there. No encoded files, but on asterisk 1.2
>> we have, it's the same file and it works.
>
> Hmm . . only one wav file. We had several. As I recall now, we
> actually installed 1.6.1.1 and upgraded. 1.6.1.1 had the old hold
> music. 1.6.1.6 has the new hold music. But I believe there are several
> files. Is that wav file valid, i.e., if you copy it to a system with a
> sound card and play it, does it play? Could it have been corrupted in
> copying or have incorrect permissions? - John
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[asterisk-users] RTP Delayed during RTCP
Hello, Has anyone encountered that when Asterisk sends RTCP messages, it stops sending RTP packets until it gets an answer? Can that be fixed? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
> I'm afraid I can't be much help as I am both a newbie and it works just > fine for me on 1.6.1.6. Of course, mine was a fresh installation. Thanks for your help, John. Mine is also a fresh installation, but now at least I know it's not a version issue. > Is there anything in the logs to give you a clue? There's absolutely nothing in the logs, and that's what surprises me. > You see the wav files but do you see the files encoded for the codecs you are using? There's only one wav file there. No encoded files, but on asterisk 1.2 we have, it's the same file and it works. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
Hello, We posted the question below yesterday, but got no answer from the community. When we checked the same behavior with Asterisk 1.2, we got the "Started music on hold, class..." message on the console, but in 1.6, we get absolutely nothing. I tried to unload and reload the moh module and everything seems normal, but Asterisk still doesn't respond in the console to the HOLD action, represented by the INVITE message. the call itself is being placed on hold and can be retrieved, but the audio file is not played and the held party hears only a silence. If anyone knows how to debug/fix it, your help would be HIGHLY appreciated. We're really stuck. Thank you all in advance. Original Message Subject: Music On Hold From: Cyprus VoIP To: Asterisk Users Mailing List - Non-Commercial Discussion Date: Tuesday, 29 September, 2009 14:31:28 > Hello, > > We need help in debugging Music On Hold on our Asterisk 1.6.1.6 > > From the SIP debug, I see that an extension sends an INVITE of the call > to the Asterisk, whenever the HOLD or Transfer buttons are pressed, but > I don't see in the console any reference to the call being placed on hold. > > When I typed "moh show files", I see the wav files of the > /var/lib/asterisk/moh folder. > > How can I debug this? > > Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music On Hold
Hello, We need help in debugging Music On Hold on our Asterisk 1.6.1.6 From the SIP debug, I see that an extension sends an INVITE of the call to the Asterisk, whenever the HOLD or Transfer buttons are pressed, but I don't see in the console any reference to the call being placed on hold. When I typed "moh show files", I see the wav files of the /var/lib/asterisk/moh folder. How can I debug this? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Versions of Asterisk 1.6
Hello, I see that there's 1.6.0.x and 1.6.1.y versions of Asterisk. Is there a clear table that describes the features and/or differences between them? Are both stable enough? Is T.38 Fax supported on both? If yes, which spandsp is supported? I saw on voip-info.org that version 6 is not supported, but this information might be outdated. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crystal Recording Interface
Hi, Is there anyone there that installed successfully the CRI package and manages to play the calls listed in the "call monitor" page? Regards. Original Message Subject: Re: [asterisk-users] Crystal Recording Interface From: Cyprus VoIP To: Asterisk Users Mailing List - Non-Commercial Discussion Date: Monday, 31 August, 2009 02:11:21 > I manage to see the calls in the monitor now, but unlike the example on > Tikal's site, I don't have the "Play Call" button next to each call, so > I can't listen to it. Could it be linked to the file name that I used > for storing the recorded calls? > > Original Message > Subject: Re: [asterisk-users] Crystal Recording Interface > From: Danny Nicholas > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Date: Friday, 28 August, 2009 18:56:46 > >> The key as far as I can see is what is in your CDR database. Once it is >> correct, everything will be as expected. This might not be a fun hill to >> climb. >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP >> Sent: Friday, August 28, 2009 10:37 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] Crystal Recording Interface >> >> I installed the asterisk add-ons (mysql,cdr), Apache MySQL and php are >> installed and running and I have the CRI web interface available, too. >> >> Seems to me I just need the extensions.conf and maybe something to do >> with CDRs to work out. >> >> Original Message >> Subject: Re: [asterisk-users] Crystal Recording Interface >> From: Danny Nicholas >> To: 'Asterisk Users Mailing List - Non-Commercial Discussion' >> >> Date: Friday, 28 August, 2009 18:22:49 >> >>> As far as I can see, the main requirements are these: >>> 1. An Apache install to install the software into >>> 2. PHP is active >>> 3. Your Asterisk uses a MYSQL or Postgres CDR. >>> 4. You have access to the database password/id. >>> >>> I'm guessing the documentation is pretty scarce since it's a >>> "one-trick-pony" GPL offering. >>> -Original Message- >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus >>> VoIP >>> Sent: Friday, August 28, 2009 9:53 AM >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: [asterisk-users] Crystal Recording Interface >>> >>> Hello all, >>> >>> >>> I download from Tikal's site the Crystal Recording Interface and >>> installed it on my Asterisk server, but there's no reference in the >>> installation instructions there regarding the necessary settings on >>> the Asterisk itself. >>> >>> >>> Is anyone using it? Any detailed explanation on the implementation of >>> that solution anywhere? >>> >>> >>> Thanks. >>> >>> >>> ___ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >>> Register Now: http://www.astricon.net >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> ___ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >>> Register Now: http://www.astricon.net >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crystal Recording Interface
I manage to see the calls in the monitor now, but unlike the example on Tikal's site, I don't have the "Play Call" button next to each call, so I can't listen to it. Could it be linked to the file name that I used for storing the recorded calls? Original Message Subject: Re: [asterisk-users] Crystal Recording Interface From: Danny Nicholas To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Date: Friday, 28 August, 2009 18:56:46 > The key as far as I can see is what is in your CDR database. Once it is > correct, everything will be as expected. This might not be a fun hill to > climb. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP > Sent: Friday, August 28, 2009 10:37 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Crystal Recording Interface > > I installed the asterisk add-ons (mysql,cdr), Apache MySQL and php are > installed and running and I have the CRI web interface available, too. > > Seems to me I just need the extensions.conf and maybe something to do > with CDRs to work out. > > Original Message > Subject: Re: [asterisk-users] Crystal Recording Interface > From: Danny Nicholas > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Date: Friday, 28 August, 2009 18:22:49 > >> As far as I can see, the main requirements are these: >> 1. An Apache install to install the software into >> 2. PHP is active >> 3. Your Asterisk uses a MYSQL or Postgres CDR. >> 4. You have access to the database password/id. >> >> I'm guessing the documentation is pretty scarce since it's a >> "one-trick-pony" GPL offering. >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP >> Sent: Friday, August 28, 2009 9:53 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: [asterisk-users] Crystal Recording Interface >> >> Hello all, >> >> >> I download from Tikal's site the Crystal Recording Interface and >> installed it on my Asterisk server, but there's no reference in the >> installation instructions there regarding the necessary settings on the >> Asterisk itself. >> >> >> Is anyone using it? Any detailed explanation on the implementation of >> that solution anywhere? >> >> >> Thanks. >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help - CDR MySQL
I already did that and now cdr_addon_mysql.so is loaded, but I still don't get anything into the database. How can I debug it? Original Message Subject: Re: [asterisk-users] Need help - CDR MySQL From: Doug Lytle To: Asterisk Users Mailing List - Non-Commercial Discussion Date: Sunday, 30 August, 2009 19:27:46 > Cyprus VoIP wrote: >> I think that the missing component is mysqlclient, but when i "yum >> update mysql", it does nothing. >> >> > > You need to make sure that mysql-devel is installed and then re-compile > add-ons > > Doug > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help - CDR MySQL
OK, Installed all the dependencies and recompiled everything.
My cdr_addon_mysql.so loads now, but still nothing is logged in the
database.
I ran this on the CLI:
localhost*CLI> cdr show status
CDR logging: enabled
CDR mode: simple
CDR output unanswered calls: yes
CDR registered backend: cdr_manager
CDR registered backend: cdr-custom
CDR registered backend: Adaptive ODBC
CDR registered backend: csv
Should there be a reference here to MySQL?
Anything else I should set?
Thanks.
Original Message
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: hh174
To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Sunday, 30 August, 2009 18:58:48
yum search mysql client
yum install 'TheClientYumHasReturnedForYourSystem'
Olivier
Cyprus VoIP a écrit :
I think that the missing component is mysqlclient, but when i "yum
update mysql", it does nothing.
Anyone know how to download the RPM? I'm using CentOS 5.3.
Thanks.
Original Message
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Pascal Bruno
To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Sunday, 30 August, 2009 18:28:35
You have to fix the dependency issues, which means install the stuff
you are missing that cdrmysql depends on so u can recompile it.
Sent from my iPod
On Aug 30, 2009, at 11:18 AM, Cyprus VoIP wrote:
Thanks. I found out that the module didn't load:
[Aug 30 20:35:59] WARNING[31906]: loader.c:371 load_dynamic_module:
Error loading module 'cdr_addon_mysql.so':
/usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared
object
file: No such file or directory
When I checked, I saw that it doesn't exist. It seems that when I
installed the addons, I didn't realize that there were some issues to
resolve first, as in the menuselect, I see that both
app_addon_sql_mysql
and cdr_addon_mysql have dependencies problems.
What should I do to resolve that?
Thanks.
Original Message
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Tilghman Lesher
To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Sunday, 30 August, 2009 17:17:59
On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote:
Hello all,
I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon,
but
without success.
Is there a proper online manual that describes all the steps to
follow
and debugging/monitoring information?
When I type in the CLI "module show", cdr_addon_mysql.so is not
listed,
although in modules.conf, I added the line "load =>
cdr_addon_mysql.so".
I also tried "preload", but it didn't change anything.
I also checked "res_mysql.conf" and "cdr_mysql.conf", and the entire
necessary data for the mysql server is there.
In "cdr_manager.conf" and "cdr.conf", I set "enabled = yes" in
"[general]".
I would appreciate any help I can get at this point, as I'm
clueless as
to what can be wrong.
Type: 'module load cdr_addon_mysql.so' and correct any errors you
see.
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Re: [asterisk-users] Need help - CDR MySQL
I think that the missing component is mysqlclient, but when i "yum update mysql", it does nothing. Anyone know how to download the RPM? I'm using CentOS 5.3. Thanks. Original Message Subject: Re: [asterisk-users] Need help - CDR MySQL From: Pascal Bruno To: Asterisk Users Mailing List - Non-Commercial Discussion Date: Sunday, 30 August, 2009 18:28:35 > You have to fix the dependency issues, which means install the stuff > you are missing that cdrmysql depends on so u can recompile it. > > Sent from my iPod > > On Aug 30, 2009, at 11:18 AM, Cyprus VoIP wrote: > >> Thanks. I found out that the module didn't load: >> >> [Aug 30 20:35:59] WARNING[31906]: loader.c:371 load_dynamic_module: >> Error loading module 'cdr_addon_mysql.so': >> /usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared >> object >> file: No such file or directory >> >> When I checked, I saw that it doesn't exist. It seems that when I >> installed the addons, I didn't realize that there were some issues to >> resolve first, as in the menuselect, I see that both >> app_addon_sql_mysql >> and cdr_addon_mysql have dependencies problems. >> >> What should I do to resolve that? >> >> Thanks. >> >> Original Message >> Subject: Re: [asterisk-users] Need help - CDR MySQL >> From: Tilghman Lesher >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> >> Date: Sunday, 30 August, 2009 17:17:59 >> >>> On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote: >>>> Hello all, >>>> >>>> I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon, >>>> but >>>> without success. >>>> >>>> Is there a proper online manual that describes all the steps to >>>> follow >>>> and debugging/monitoring information? >>>> >>>> When I type in the CLI "module show", cdr_addon_mysql.so is not >>>> listed, >>>> although in modules.conf, I added the line "load => >>>> cdr_addon_mysql.so". >>>> I also tried "preload", but it didn't change anything. >>>> >>>> I also checked "res_mysql.conf" and "cdr_mysql.conf", and the entire >>>> necessary data for the mysql server is there. >>>> >>>> In "cdr_manager.conf" and "cdr.conf", I set "enabled = yes" in >>>> "[general]". >>>> >>>> I would appreciate any help I can get at this point, as I'm >>>> clueless as >>>> to what can be wrong. >>> Type: 'module load cdr_addon_mysql.so' and correct any errors you >>> see. >>> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help - CDR MySQL
Thanks. I found out that the module didn't load: [Aug 30 20:35:59] WARNING[31906]: loader.c:371 load_dynamic_module: Error loading module 'cdr_addon_mysql.so': /usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared object file: No such file or directory When I checked, I saw that it doesn't exist. It seems that when I installed the addons, I didn't realize that there were some issues to resolve first, as in the menuselect, I see that both app_addon_sql_mysql and cdr_addon_mysql have dependencies problems. What should I do to resolve that? Thanks. Original Message Subject: Re: [asterisk-users] Need help - CDR MySQL From: Tilghman Lesher To: Asterisk Users Mailing List - Non-Commercial Discussion Date: Sunday, 30 August, 2009 17:17:59 > On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote: >> Hello all, >> >> I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon, but >> without success. >> >> Is there a proper online manual that describes all the steps to follow >> and debugging/monitoring information? >> >> When I type in the CLI "module show", cdr_addon_mysql.so is not listed, >> although in modules.conf, I added the line "load => cdr_addon_mysql.so". >> I also tried "preload", but it didn't change anything. >> >> I also checked "res_mysql.conf" and "cdr_mysql.conf", and the entire >> necessary data for the mysql server is there. >> >> In "cdr_manager.conf" and "cdr.conf", I set "enabled = yes" in "[general]". >> >> I would appreciate any help I can get at this point, as I'm clueless as >> to what can be wrong. > > Type: 'module load cdr_addon_mysql.so' and correct any errors you see. > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help - CDR MySQL
Hello all, I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon, but without success. Is there a proper online manual that describes all the steps to follow and debugging/monitoring information? When I type in the CLI "module show", cdr_addon_mysql.so is not listed, although in modules.conf, I added the line "load => cdr_addon_mysql.so". I also tried "preload", but it didn't change anything. I also checked "res_mysql.conf" and "cdr_mysql.conf", and the entire necessary data for the mysql server is there. In "cdr_manager.conf" and "cdr.conf", I set "enabled = yes" in "[general]". I would appreciate any help I can get at this point, as I'm clueless as to what can be wrong. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crystal Recording Interface
I installed the asterisk add-ons (mysql,cdr), Apache MySQL and php are installed and running and I have the CRI web interface available, too. Seems to me I just need the extensions.conf and maybe something to do with CDRs to work out. Original Message Subject: Re: [asterisk-users] Crystal Recording Interface From: Danny Nicholas To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Date: Friday, 28 August, 2009 18:22:49 > As far as I can see, the main requirements are these: > 1. An Apache install to install the software into > 2. PHP is active > 3. Your Asterisk uses a MYSQL or Postgres CDR. > 4. You have access to the database password/id. > > I'm guessing the documentation is pretty scarce since it's a > "one-trick-pony" GPL offering. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP > Sent: Friday, August 28, 2009 9:53 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Crystal Recording Interface > > Hello all, > > > I download from Tikal's site the Crystal Recording Interface and > installed it on my Asterisk server, but there's no reference in the > installation instructions there regarding the necessary settings on the > Asterisk itself. > > > Is anyone using it? Any detailed explanation on the implementation of > that solution anywhere? > > > Thanks. > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Crystal Recording Interface
Hello all, I download from Tikal's site the Crystal Recording Interface and installed it on my Asterisk server, but there's no reference in the installation instructions there regarding the necessary settings on the Asterisk itself. Is anyone using it? Any detailed explanation on the implementation of that solution anywhere? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] onnecting two asterisk using B410p BRI cards
I just plug the junper in NT mode with no success. VoipCrazy 2009/8/15 Paul Hales : > > Use a standard network cable - but you have to activate the 'terminate' > jumper on the NT end. > > - Also, the new BRI stuff in dahdi is much easier to work with than misdn. > > PaulH > > > voip crazy wrote: >> Hello all, >> >> I'm trying to conect two asterisk servers using two B410p Digium >> cards. One card on each server. I just setting up the first BRI port >> on server A as nt_ptp and the first BRI port on server B as te_ptp. >> I use an ethernet wire to connect the first port of server A (nt_ptp) >> with the first port on server B (te_ptp) but the port light cotinues >> blinking on red on both sides once the cable was pluged. Then I use an >> isdn crossover wire with this king of schema and the lights get >> blinking red again. >> >> Tx+ 3 --+ +- 3 >> . X >> Rx+ 4 --+ +- 4 >> . >> Tx- 5 --+ +--5 >> . X >> Rx- 6 --+ +--6 >> >> In both servers when I do in asterisk CLI "misdn shos stacks", the >> port one on each machine shows >> >> Server A: >> >> BEGIN STACK_LIST: >> * Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 >> >> >> Server B: >> >> BEGIN STACK_LIST: >> * Port 1 Type NT Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 >> >> Which kind of cable should I use? >> Why both in ports L1Link is failed? >> How could I solve that? >> >> Any clue will be welcomed. >> >> Thanks in advance. >> >> VoipCrazy. >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] onnecting two asterisk using B410p BRI cards
Hello all, I'm trying to conect two asterisk servers using two B410p Digium cards. One card on each server. I just setting up the first BRI port on server A as nt_ptp and the first BRI port on server B as te_ptp. I use an ethernet wire to connect the first port of server A (nt_ptp) with the first port on server B (te_ptp) but the port light cotinues blinking on red on both sides once the cable was pluged. Then I use an isdn crossover wire with this king of schema and the lights get blinking red again. Tx+ 3 --+ +- 3 .X Rx+ 4 --+ +- 4 . Tx- 5 --+ +--5 .X Rx- 6 --+ +--6 In both servers when I do in asterisk CLI "misdn shos stacks", the port one on each machine shows Server A: BEGIN STACK_LIST: * Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 Server B: BEGIN STACK_LIST: * Port 1 Type NT Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 Which kind of cable should I use? Why both in ports L1Link is failed? How could I solve that? Any clue will be welcomed. Thanks in advance. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] INVITE Privacy Information
That's exactly what I ended up doing: SIPAddHeader(Remote-Party-ID: \;privacy=full\;screen=yes) Note the \ before each ; It wouldn't put anything behind the ; without them. Thanks. Original Message Subject: Re: [asterisk-users] INVITE Privacy Information From: Philipp Kempgen To: Asterisk Users Mailing List - Non-Commercial Discussion Date: Tuesday, 28 July, 2009 14:10:55 > Cyprus VoIP schrieb: >> I ran into this problem: When I change the CALLERID(num and name) to >> anonymous, they are also changed in the RPID line and not only in the From. > > OK. I'd try to set sendrpid=no in sip.conf and then add a > Remote-Party-ID header in the dialplan. > SIPAddHeader(Remote-Party-ID: ;screen=yes;privacy=full); > I have no idea if Asterisk will let you do that even if sendrpid > is disabled. > >> Original Message >> Subject: Re: [asterisk-users] INVITE Privacy Information >> From: Philipp Kempgen >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> >> Date: Monday, 27 July, 2009 17:16:45 >> >>> Cyprus VoIP schrieb: > >>> These are the sip headers I need to add to the INVITE: >>>> /P-Asserted-Identity: >>> SIPAddHeader(P-Asserted-Identity: ); // RFC 3325 >>> >>>> Remote-Party-ID: >>>> ;party=calling;screen=yes;privacy=full >>> Enable >>> sendrpid=yes ; If Remote-Party-ID should be sent >>> in the [general] section in sip.conf. >>> SetCallerPres(prohib_passed_screen); > >>>> And I need to change the "From" to >>>> "/"Anonymous"/" and to remove the "*67" >>>> prefix from the "INVITE" and "To" lines. >>> Set(CALLERID(num)=anonymous); // RFC 2543 >>> Set(CALLERID(name)=Anonymous); > > > Philipp Kempgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] INVITE Privacy Information
Thank you Philipp for your help.
I ran into this problem: When I change the CALLERID(num and name) to
anonymous, they are also changed in the RPID line and not only in the From.
This is the script:
exten => _*67.,1,SIPAddHeader(Privacy: id);
exten => _*67.,2,Set(CALLERPRES()=prohib_passed_screen);
exten => _*67.,3,SIPAddHeader(P-Asserted-Identity:
);
exten => _*67.,4,Set(CALLERID(num)=anonymous)
exten => _*67.,5,Set(CALLERID(name)=Anonymous);
exten => _*67.,6,Dial(SIP/${EXTEN:3...@10.10.10.10)
... and this is the result:
INVITE sip:0011223...@10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.20:5060;branch=z9hG4bK3031c135
Max-Forwards: 70
From: "Anonymous" ;tag=as6d95f136
To:
Contact:
Call-ID: 1732ae8f6581677c4a0d46360b102...@10.10.10.20
CSeq: 102 INVITE
User-Agent: Asterisk-PBX
Remote-Party-ID: "Anonymous"
;privacy=full;screen=yes
Date: Tue, 28 Jul 2009 09:00:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
P-Asserted-Identity:
Privacy: id
Original Message
Subject: Re: [asterisk-users] INVITE Privacy Information
From: Philipp Kempgen
To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Monday, 27 July, 2009 17:16:45
> Cyprus VoIP schrieb:
>> I would like to use Asterisk to add/modify SIP headers in the INVITE
>> message, to include Privacy information, if the INVITE includes a *67
>> prefix (or another predefined prefix).
>>
>> That's an example of the INVITE I get:
>> /INVITE sip:*6700112233...@192.168.1.100 SIP/2.0
>> From: "123456789";tag=3
>> To:
>> /
>> These are the sip headers I need to add to the INVITE:
>> /P-Asserted-Identity:
>
> SIPAddHeader(P-Asserted-Identity: ); // RFC 3325
>
>> Remote-Party-ID:
>> ;party=calling;screen=yes;privacy=full
>
> Enable
> sendrpid=yes ; If Remote-Party-ID should be sent
> in the [general] section in sip.conf.
> SetCallerPres(prohib_passed_screen);
>
>> Privacy: id/
>
> SIPAddHeader(Privacy: id); // RFC 3325, RFC 3323
>
>> And I need to change the "From" to
>> "/"Anonymous"/" and to remove the "*67"
>> prefix from the "INVITE" and "To" lines.
>
> Set(CALLERID(num)=anonymous); // RFC 2543
> Set(CALLERID(name)=Anonymous);
>
>
> Philipp Kempgen
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[asterisk-users] INVITE Privacy Information
Hello all, I would like to use Asterisk to add/modify SIP headers in the INVITE message, to include Privacy information, if the INVITE includes a *67 prefix (or another predefined prefix). That's an example of the INVITE I get: /INVITE sip:*6700112233...@192.168.1.100 SIP/2.0 From: "123456789";tag=3 To: / These are the sip headers I need to add to the INVITE: /P-Asserted-Identity: Remote-Party-ID: ;party=calling;screen=yes;privacy=full Privacy: id/ And I need to change the "From" to "/"Anonymous"/" and to remove the "*67" prefix from the "INVITE" and "To" lines. How can I do it? Thanks. Thierry ___ -- 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] Manipulating REGISTER messages
Hello, I would like to add SIP headers to the REGISTER messages Asterisk (1.6) sends to an external proxy. Also, I want to be able to reorder the lines. Is it possible? If yes, how? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Printing faxes
Hello list, I have an asterisk / hylafax / iaxmodem configured in one machine. All is working nicely. Now I need the fax to be print when arriving. ¿Anybody have this feature implementing in their systems? ¿How is the best way to get that? Any clue will be welcomed. Thanks. VoipCrazy ___ -- 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] Webcall app needed
Hello all, I need to configure an application which let me to call from a web page. Someone has experience using apps to make webcalls? Which software do you use? Thanks. VoipCrazy. ___ -- 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] Hylafax asterisk iaxmodem problem
Hello all, I have an asterisk box running in a customer with Hylafax, iaxmodem, asterisk 1.2.18. The service can receive faxes, from a lot of fax machines, but there are a couple of them that asterisk Hylafax cannot complete. This calls arrive the asterisk box, asterisk detect that this calls are fax, asterisk answer the call, and then Hangup the call. But hylafax do not receive nothing. When I run "zap show channel 1", on the asterisk CLI. The outpuit shows, File Descriptor: 20 Span: 2 Extension: Dialing: no Context: from-zaptel Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: Real: Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook Why some faxes do not get received? What could be wrong? Any clue wil be welcomed. Thanks in advanced. VoipCrazy. ___ -- 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] WebCall application
Hello list, Does anybody know any free WebCall solution to let our customer call us directly via our web site? Any clue will be welcomed. Thanks. VoipCrazy ___ -- 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] Asterisk Queue question
When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. >show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy "Fewest Calls" working for a couple of mouths, and a new agent has been added this week in the queue and he is receiving all the incomings calls. How could I solve that? Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] B410p question
Hello list, I have got an asterisk box installed working ok with an b410p card to make and receive isdn calls. All works ok, but when a call is answer and the person starts to speak, always I can ear a "beep" during the call. This beep is ear some times in about 30 seconds between each beep. Pasted bellow I send /etc/misdn-init.conf and /etc/asterisk/misdn.conf Any clue will be apreciated. Thanks. VoipCrazy - My /etc/misdn-init.conf - # # Configuration file for your misdn hardware # # Usage: /usr/sbin/misdn-init start|stop|restart|config|scan|help # # # Card Settings # # Syntax: card=,[,...] # # count your cards beginning with 1 # either 0x1,0x4 or 0x8 for your hfcmulti hardware, # or the name of your card driver module. # ulaw - uLaw (instead of aLaw) # dtmf - enable DTMF detection on all B-channels # # pcm_slave - set PCM bus into slave mode #If you have a set of cards, all wired via PCM. Set #all cards into pcm_slave mode and leave one out. #The left card will automatically be Master. # # ignore_pcm_frameclock - this can be set in conjunction with # pcm_slave. If this card has a # PCI Bus Position before the Position # of the Master, then this port cannot # yet receive a frameclock, so it must # ignore the pcm frameclock. # # rxclock- use clocking for pcm from ST Port # crystalclock - use clocking for pcm from PLL (genrated on board) # watchdog - This dual E1 Board has a Watchdog for #transparent mode # # card=1,0x4 # # Port settings # # Syntax: =[,...] # #te_ptp - TE-Mode, PTP # te_ptmp - TE-Mode, PTMP # te_capi_ptp - TE-Mode (capi), PTP # te_capi_ptmp- TE-Mode (capi), PTMP # nt_ptp - NT-Mode, PTP # nt_ptmp - NT-Mode, PTMP # port that should be considered # #te_ptmp=1,2,3,4 #te_ptmp=1,2 te_ptp=1,2,3,4 # # Port Options # # Syntax: option=,[,...] # # master_clock - use master clock for this S/T interface # (only once per chip, only for HFC 8/4) # optical - optical (only HFC-E1) # los - report LOS (only HFC-E1) # ais - report AIS (only HFC-E1) # slip - report SLIP (only HFC-E1) # nocrc4- turn off crc4 mode use double frame instead # (only HFC-E1) # # The master_clock option is essential for retrieving and transmitting # faxes to avoid failures during transmission. It tells the driver to # synchronize the Card with the given Port which should be a TE Port and # connected to the PSTN in general. # option=1,master_clock #option=2,ais,nocrc4 #option=3,optical,los,ais,slip # # General Options for your hfcmulti hardware # # poll= # #Only one poll value must be given for all cards. #Give the number of samples for each fifo process. #By default 128 is used. Decrease to reduce delay, increase to #reduce cpu load. If unsure, don't mess with it!!! #Valid is 32, 64, 128, 256. # # dsp_poll= # This is the poll option which is used by mISDN_dsp, this might # differ from the one given by poll= for the hfc based cards, since # they can only use multiples of 32, the dsp_poll is dependant on # the kernel timer setting which can be found in the CPU section # in the kernel config. Defaults are there either 100Hz, 250Hz # or 1000Hz. If your setting is either 1000 or 250 it is compatible # with the poll option for the hfc chips, if you have 100 it is # different and you need here a multiple of 80. # The default is to have no dsp_poll option, then the dsp itself # finds out which option is the best to use by itself # # pcm= # #Give the id of the PCM bus. All PCM busses with the same ID #are expected to be connected and have equal slots. #Only one chip of the PCM bus must be master, the others slave. # # debug= # #Enable debugging (see hfc_multi.h for debug options). # # dsp_options= # # set this to 2 and you'll have software bridging instead of # hardware bridging. # # # dtmfthreshold= # # Here you can tune the sensitivity of the dtmf tone recognizer. # # timer=<1|0> # # set this to 1 if you want hfcmulti to register at ztdummy (zaptel) # and provide a 1khz timing source for it. This makes it possible # to have an accurate timing source f
[asterisk-users] Sip Header Help
Dear List: I need to make a sip phone (spa942) answer a call but the phone must no ring. The user only has to show the callerId on the phone screen without any sound. How could I make that in asterisk? I tried to use Sip headers but I do not know how must I say the phone don't ring when received, only shows the callerID of the call. How could I do that with sip header? Which sip header should I send the phone to change the callerID of the call? Do you know any other way to ghet that. Any clue will be wellcomed. Thanks for your answer. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dundi and regcontext
According to Your description this is a phone problem. Asterisk behaves as its expected. post your dundi.conf to dig more in to this. regards rama On Wed, Sep 24, 2008 at 9:52 PM, ronald ramos <[EMAIL PROTECTED]>wrote: > hi, > > when a user register on my asterisk i can see it adding Noop for that > extension, but after awhile i won't see it anymore: > > what are the reasons for it being removed on the dynamic context? > one thing i found when i unregister it's removed. > > dialplan show myregcontext > [ Context 'myregcontext' created by 'SIP' ] > '100500' => 1. Noop(100500) [SIP] > '112802' => 1. Noop(112802) [SIP] > > -= 2 extensions (2 priorities) in 1 context. =- > > [ Context 'pfingobizsip' created by 'SIP' ] > > -= 0 extensions (0 priorities) in 1 context. =- > > my prob is when it's removed dundi cant find it anymore so a user calling > from server 1 cannot call user that is in server 2. > > i've set re-registration to very low (1 minute) to monitor if my phone > re-register and to see if it will be added again on the regcontext. > but i don't even see it unregistering after 1 minute i only unregistering > when i am using x-lite and closing x-lite, i dont see x-lite re-registering > if i just leave the softphone open. any idea? > > regards, > ron > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dundi Help
hi gior,
If i understand correctly your setup would be like below.
A is the dundi serer
B is one pbx
C is one pbx
B and C dundi.conf contain the entity detials of A.
Either for C or B we can place calls to the extensions registered on the
other server.
When C extension make call to B extension call go through A and reach the B.
If you can provide the details of dundi.conf and the extension.conf it would
be very help full.
to dig into the issue.
could you able to do the dundi lookup ..?
regards
On Wed, Sep 10, 2008 at 3:36 PM, Giorgio Incantalupo <
[EMAIL PROTECTED]> wrote:
> Hi tecnocrat,
>
> I 'm trying to setup a Dundi system like yours (one lookup server and 2
> pbx servers). I searched on internet for some docs but found a lot of
> stuff explaining only a part of the problem and no good example at all
> (there's a Richardson doc in internet which can help to start).
> I tried to do it myself so I generated the two keys (pri and pub) for
> each server with their own hostname then I copied:
> - .121 keys to the other two servers (.137 and .204)
> - .137 keys to .121
> - .204 keys to .121
>
> Let me know how if it works.
>
> Giorgio Incantalupo
>
>
> technocrat voip wrote:
> > Hello All,
> >
> > Iam trying to achive a simple load balancing with dundi.
> >
> > Here i have three asterisk boxes like below.
> >
> >
> > *.*.*.121 which is the dundi server
> >
> > *.*.*.137 "A" Peer which has the 1000 phone registerd to it
> >
> > *.*.*.204 "B" Peer which has the 200 phone registered to it.
> >
> > The expected behavior of my setup is once i dial from 1000 phone it
> > has to goto "B" peer using the ".121" dundi server.
> >
> > Iam getting confused with the public key / private key stuff here.
> >
> > Iam using the "astgenkey -n " command to generate them .
> >
> > Can any body help me by explaining , what keys i have to generate on
> > each server and which keys i need to copy to which server .
> >
> > regards
> >
> >
> >
> > All the conf file stuff is is like below
> >
> > *.*.*.137
> >
> > iax.conf
> >
> > [priv]
> > type=friend
> > dbsecret=dundi/secret
> > context=incomingdundi
> >
> >
> > dundi.conf
> >
> > [mappings]
> > priv =>
> > sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
> >
> >
> > [00:0F:E2:76:4B:33]
> > model = symmetric
> > host = *.*.*.21
> > inkey = dundincomingseven
> > outkey = dundiseven
> > include = priv
> > permit = priv
> > qualify = yes
> > order = primary
> >
> >
> > *.*.*.204
> >
> > iax.conf
> >
> > [priv]
> > type=friend
> > dbsecret=dundi/secret
> > context=incomingdundi
> >
> > dundi.conf
> >
> > [mappings]
> > priv =>
> > sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
> > ;
> >
> > [00:0F:E2:76:4B:33]
> > model = symmetric
> > host = *.*.*.21
> > inkey = dundiincomingfour
> > outkey = dundifour
> > include = priv
> > permit = priv
> > qualify = yes
> > order = primary
> >
> > *.*.*.21
> >
> > iax.conf
> > [priv]
> > type=user
> > context=local-custom
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=gsm
> >
> >
> > dundi.conf
> >
> >
> > [mappings]
> > priv => dundi-priv-localcustom,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER}
> >
> > [00:11:0A:34:29:57]
> > model = symmetric
> > host = *.*.*.137
> > inkey = dundi
> > outkey = dundi
> > include = priv
> > permit = priv
> > qualify = yes
> > order = primary
> >
> > [00:11:0A:34:29:43]
> > model = symmetric
> > host = *.*.*.204
> > inkey = dundi
> > outkey = dundi
> > include = priv
> > permit = priv
> > qualify = yes
> > order = primary
> > .
> >
> >
> >
> >
> >
> >
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
>
> _
> Giorgio Incantalupo, mailto:[EMAIL PROTECTED]
> FG&A srl - http://www.fgasoftware.com -
> [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
> Tel: 02997663.14, Fax: 0291390172
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] Dundi Help
Hello All,
Iam trying to achive a simple load balancing with dundi.
Here i have three asterisk boxes like below.
*.*.*.121 which is the dundi server
*.*.*.137 "A" Peer which has the 1000 phone registerd to it
*.*.*.204 "B" Peer which has the 200 phone registered to it.
The expected behavior of my setup is once i dial from 1000 phone it has to
goto "B" peer using the ".121" dundi server.
Iam getting confused with the public key / private key stuff here.
Iam using the "astgenkey -n " command to generate them .
Can any body help me by explaining , what keys i have to generate on each
server and which keys i need to copy to which server .
regards
All the conf file stuff is is like below
*.*.*.137
iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=incomingdundi
dundi.conf
[mappings]
priv => sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:0F:E2:76:4B:33]
model = symmetric
host = *.*.*.21
inkey = dundincomingseven
outkey = dundiseven
include = priv
permit = priv
qualify = yes
order = primary
*.*.*.204
iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=incomingdundi
dundi.conf
[mappings]
priv => sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
;
[00:0F:E2:76:4B:33]
model = symmetric
host = *.*.*.21
inkey = dundiincomingfour
outkey = dundifour
include = priv
permit = priv
qualify = yes
order = primary
*.*.*.21
iax.conf
[priv]
type=user
context=local-custom
disallow=all
allow=ulaw
allow=alaw
allow=gsm
dundi.conf
[mappings]
priv => dundi-priv-localcustom,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER}
[00:11:0A:34:29:57]
model = symmetric
host = *.*.*.137
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary
[00:11:0A:34:29:43]
model = symmetric
host = *.*.*.204
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary
.
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Re: [asterisk-users] Gateway errors
Thank you Hatem, I will try it now Thanks VoipCrazy 2008/9/2 hatem moiz <[EMAIL PROTECTED]>: > you can do the following in sip .conf file > > register => username:[EMAIL PROTECTED] > > and after that write the configuration for the user: > > [ user ] > username = > host = > qualify = > secret = > > and so on, do this in the first of sip.conf file > > Best Regards > > On Mon, Sep 1, 2008 at 11:32 AM, voip crazy <[EMAIL PROTECTED]> wrote: >> >> Hatem, >> >> I cannot understan exactly what you told me. >> Could you try to explain that in other words. Better if you could post >> an example of this SIP trunk. >> >> thanks in advance. >> >> Voip Crazy >> >> >> >> 2008/9/1 hatem moiz <[EMAIL PROTECTED]>: >> > Asterisk is looking for a SIP trunk if you have recorded the usage of >> > SIP >> > trunks all it need is to find 1 SIP trunk, >> > >> > To fix your problem make a local sip trunk i mean sip trunk to 127.0.0.1 >> > and >> > make sure that it is the first one in sip.conf file. OR you can make a >> > sip >> > >> > trunk to ATA in the same lan and also be sure that it is the first trunk >> > in >> > sip.conf . >> > >> > On Mon, Sep 1, 2008 at 9:58 AM, Igor Hernandez <[EMAIL PROTECTED]> wrote: >> >> >> >> Thats strange, have you checked that you're not having issues with your >> >> router? Can you reach all the boxes in your lan while you are >> >> experiencing this downtime? >> >> >> >> voip crazy wrote: >> >> > When I say extensions, I say extensions in the lan not in wan >> >> > >> >> > Thanks. >> >> > >> >> > VoipCrazy. >> >> > >> >> > 2008/9/1 Igor Hernandez <[EMAIL PROTECTED]>: >> >> >> Hello, >> >> >> >> >> >> By people do you mean people in the lan or external users? >> >> >> >> >> >> Regards, >> >> >> >> >> >> -- >> >> >> Igor Hernandez >> >> >> Escape Communications >> >> >> http://www.escapetel.com >> >> >> >> >> >> >> >> >> voip crazy wrote: >> >> >>> Hello list, >> >> >>> >> >> >>> I have an asterisk instalation with a bad internet connection cause >> >> >>> this connection is down sometimes. >> >> >>> When the connection is down and asterisk cannot get internet >> >> >>> connection. All the extensions log out from the asterisk machine, >> >> >>> and >> >> >>> nobody can make any call. >> >> >>> >> >> >>> ¿Why if internet connection is down asterisk stops working >> >> >>> correctly? >> >> >>> ¿How could I solve that? >> >> >>> >> >> >>> Thansk. >> >> >>> >> >> >>> VoipCrazy >> >> >>> ___ >> >> >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com >> >> >>> -- >> >> >>> >> >> >>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> >> >>> Register Now: http://www.astricon.net >> >> >>> >> >> >>> asterisk-users mailing list >> >> >>> To UNSUBSCRIBE or update options visit: >> >> >>>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> >> >> >> >> >> ___ >> >> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com >> >> >> -- >> >> >> >> >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> >> >> Register Now: http://www.astricon.net >> >> >> >> >> >> asterisk-users mailing list >> >> >> To UNSUBSCRIBE or update options visit: >> >> >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > ___ >> >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> > >> >> > AstriCon 2008
[asterisk-users] Asterisk 1.6 beta
Hello users, Can you share with me your experiences with Asterisk 1.6? Is it stable enough for commercial service? Thanks. > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Hatem, I cannot understan exactly what you told me. Could you try to explain that in other words. Better if you could post an example of this SIP trunk. thanks in advance. Voip Crazy 2008/9/1 hatem moiz <[EMAIL PROTECTED]>: > Asterisk is looking for a SIP trunk if you have recorded the usage of SIP > trunks all it need is to find 1 SIP trunk, > > To fix your problem make a local sip trunk i mean sip trunk to 127.0.0.1 and > make sure that it is the first one in sip.conf file. OR you can make a sip > > trunk to ATA in the same lan and also be sure that it is the first trunk in > sip.conf . > > On Mon, Sep 1, 2008 at 9:58 AM, Igor Hernandez <[EMAIL PROTECTED]> wrote: >> >> Thats strange, have you checked that you're not having issues with your >> router? Can you reach all the boxes in your lan while you are >> experiencing this downtime? >> >> voip crazy wrote: >> > When I say extensions, I say extensions in the lan not in wan >> > >> > Thanks. >> > >> > VoipCrazy. >> > >> > 2008/9/1 Igor Hernandez <[EMAIL PROTECTED]>: >> >> Hello, >> >> >> >> By people do you mean people in the lan or external users? >> >> >> >> Regards, >> >> >> >> -- >> >> Igor Hernandez >> >> Escape Communications >> >> http://www.escapetel.com >> >> >> >> >> >> voip crazy wrote: >> >>> Hello list, >> >>> >> >>> I have an asterisk instalation with a bad internet connection cause >> >>> this connection is down sometimes. >> >>> When the connection is down and asterisk cannot get internet >> >>> connection. All the extensions log out from the asterisk machine, and >> >>> nobody can make any call. >> >>> >> >>> ¿Why if internet connection is down asterisk stops working correctly? >> >>> ¿How could I solve that? >> >>> >> >>> Thansk. >> >>> >> >>> VoipCrazy >> >>> ___ >> >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >>> >> >>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> >>> Register Now: http://www.astricon.net >> >>> >> >>> asterisk-users mailing list >> >>> To UNSUBSCRIBE or update options visit: >> >>>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> >> >> ___ >> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> >> Register Now: http://www.astricon.net >> >> >> >> asterisk-users mailing list >> >> To UNSUBSCRIBE or update options visit: >> >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > ___ >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> > >> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> > Register Now: http://www.astricon.net >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> >http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez <[EMAIL PROTECTED]>: > Hello, > > By people do you mean people in the lan or external users? > > Regards, > > -- > Igor Hernandez > Escape Communications > http://www.escapetel.com > > > voip crazy wrote: >> Hello list, >> >> I have an asterisk instalation with a bad internet connection cause >> this connection is down sometimes. >> When the connection is down and asterisk cannot get internet >> connection. All the extensions log out from the asterisk machine, and >> nobody can make any call. >> >> ¿Why if internet connection is down asterisk stops working correctly? >> ¿How could I solve that? >> >> Thansk. >> >> VoipCrazy >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Igor, From asterisk, when internet is down I can ping all extensions. The same occurs in others instalations, when the internet is down, my lical extensions log off from asterisk. VoipCrazy 2008/9/1 Igor Hernandez <[EMAIL PROTECTED]>: > Thats strange, have you checked that you're not having issues with your > router? Can you reach all the boxes in your lan while you are > experiencing this downtime? > > voip crazy wrote: >> When I say extensions, I say extensions in the lan not in wan >> >> Thanks. >> >> VoipCrazy. >> >> 2008/9/1 Igor Hernandez <[EMAIL PROTECTED]>: >>> Hello, >>> >>> By people do you mean people in the lan or external users? >>> >>> Regards, >>> >>> -- >>> Igor Hernandez >>> Escape Communications >>> http://www.escapetel.com >>> >>> >>> voip crazy wrote: >>>> Hello list, >>>> >>>> I have an asterisk instalation with a bad internet connection cause >>>> this connection is down sometimes. >>>> When the connection is down and asterisk cannot get internet >>>> connection. All the extensions log out from the asterisk machine, and >>>> nobody can make any call. >>>> >>>> ¿Why if internet connection is down asterisk stops working correctly? >>>> ¿How could I solve that? >>>> >>>> Thansk. >>>> >>>> VoipCrazy >>>> ___ >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>> >>>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >>>> Register Now: http://www.astricon.net >>>> >>>> asterisk-users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>>http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> ___ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >>> Register Now: http://www.astricon.net >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gateway errors
Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outgoing calls
Hello list, How could I limit the outgoing calls for one trunks easily? Thanks VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco vs Asterisk
Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? Thanks. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dimensioning
Maybe 400 calls at one time. By the momento there aren`t voip trunks maybe in the future. About cluster, Which cluster solution will could be good option? Which solution could I use to do load balancing between two asterisk machines? Thanks again. Voipcrazy 2008/7/9 Tom Moore <[EMAIL PROTECTED]>: > How many calls do you expect to be going at one time? > Do you have any sip trunks for the users to call out on? Unless this ratio > really works for you I'm not sure a 15 to 1 ratio works for most people. > I wouldn't just depend on a single server for this purpose. > I'll leave it to the cluster guys to describe the ideal setup you should > use. > I have an idea of how I might do it, but I wouldn't want to get it wrong. > > Tom > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of voip crazy > Sent: Wednesday, July 09, 2008 3:01 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Asterisk dimensioning > > Hello all, > > I need to install asterisk for 900 sip users with 2 PRI ports. > It is posible to handle this number of calls/extensions with only one > asterisk machine? > Which is the best way to install that? two asterisk with openser. One > asterisk with openser . > Is it necesary run a SER server on this enviroment? > > Any clue will be welcomed. > > Thanks in advance. > > VoipCrazy > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk dimensioning
Hello all, I need to install asterisk for 900 sip users with 2 PRI ports. It is posible to handle this number of calls/extensions with only one asterisk machine? Which is the best way to install that? two asterisk with openser. One asterisk with openser . Is it necesary run a SER server on this enviroment? Any clue will be welcomed. Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and polycom provisioning
Hello friends, I am using the asterisk-1.6.2 , i use the gui also. I use the polycom provisioning. Now my requirement is to allow the phone to upload the log files etc to the asterisk machine. As i see now when it queries to upload the file like below T 10.231.109.206:1037 -> 10.231.109.59:80 [AP] PUT /phoneprov/0004f2184c15-app.log HTTP/1.1. Host: 10.231.109.59. Accept: */*. User-Agent: FileTransport PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084. Content-Length: 10254. Expect: 100-continue. . T 10.231.109.59:80 -> 10.231.109.206:1037 [AP] HTTP/1.1 501 Not Implemented. Can any body say is there any configuration in http.conf so that i can allow the polycom to upload this log file on the server ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Removing voicemail messages
Hello, I want to create an script which remove all the old voicemail messages. I make a simple Bash script to delete all the new messages for the extension 100. Something like, rm /var/spool/asterisk/voicemail/defaul/100/INBOX Should I update any index file or something after reemove them? Thanks in advance VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager proxy
Hello all, Some one is using asterisk and queuemetrics connected via astmanproxy? How about your experience? Which proxy do you use in this kind of connection? In my instalation asterisk and Queuemetrics are installed on diferent machines and I want to avoid manager problems Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
