[asterisk-users] sip trunk, parsing DID
Hello,
I am using a Swiss VoIP provider called sipcall. They have what they
call a SIP trunk, and it is less expensive than individual accounts. From
Asterisk's point of view, this is just a regular SIP account, which
can however receive and send calls from multiple numbers. I just migrated
from individual SIP accounts terminated on my Asterisk to one single
SIP trunk.
It works perfectly (in and out). For outgoing calls, it's just
sufficient to set CALLERID(num) to the appropriate number you want the
call to originate from (easy!).
For incoming calls, here is an example SIP message, with MY_IP, SIPCALL_IP,
DEST_NUMBER AND SRC_NUMBER replacing the actual values:
INVITE sip:s@MY_IP:5060 SIP/2.0
Via: SIP/2.0/UDP SIPCALL_IP:5060;branch=z9hG4bK3ee1k92090iihapdm420.1
Max-Forwards: 67
Contact:
To:
From: ;tag=hy4fwr752woo42uj.o
Call-ID: 1663976908-326811297@1~1o
CSeq: 867 INVITE
Expires: 300
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE,
OPTIONS, UPDATE
Content-Disposition: session
Content-Type: application/sdp
User-Agent: PortaSIP
h323-conf-id: 3912070954-288423879-3678105731-1479596074
cisco-GUID: 3912070954-288423879-3678105731-1479596074
Content-Length: 262
Since it looks that only the To: header contains the real destination number,
and debugging shows that it is not copied in ${CALLERID(all)} nor ${EXTEN}, I
had to revert to this hack, which works great:
exten => s,1,Log(NOTICE, Incoming call from sipcall-trunk ${CALLERID(all)}
to ${EXTEN} DID ${SIP_HEADER(To)})
exten => s,n,Set(DID=${SIP_HEADER(To):})
exten => s,n,Set(DID=${DID:5:11})
exten => s,n,Log(NOTICE, Parsed DID: ${DID})
exten => s,n,Goto(sipcall-trunk,sipcall-${DID},1)
exten => s,n,Hangup()
I then have individual sipcall-NUMBER handling the actions for the individual
numbers.
Is there a simpler way? Is there a safer way (check that DID only contains
numbers, e.g.?)
Thank you for any ideas or pointers.
--
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[asterisk-users] Asterisk/SIP Issue - Long Shot
Hi, This is a bit of a long shot and I don't have much information on what is actually happening... Our production Asterisk system: ~2,000 SIP handsets, (2) Digium TE220s, Asterisk 1.6.2.18, RHEL 5 x86_64 Every few weeks, or few months, or X amount of time, the SIP portion of Asterisk seems to hang/die. Nothing abnormal in the typical Asterisk logs; the rest of Asterisk still seems to function fine -- incoming DAHDI calls come in and are queued in the various queues. 'sip set debug on' gives NO messages/errors/information at all; 'module reload chan_sip.so' does nothing (never seems to unload/load it), again nothing in the logs. It just seems like the chan_sip.so (or everything SIP) just hangs -- no errors related to SIP; when Asterisk is in this state we are unable to stop it gracefully -- it always requires a kill -9. The times this happen seem totally random -- this issue has persisted through several different versions of Asterisk 1.6.2.x branch. Sometimes its weeks between occurrences, other times its months. This Asterisk host easily processes several thousand calls per day. Any ideas or if anyone could at least point us in the right direction would be greatly appreciated. When this does happen, and we need to intervene, we try to poke around for a few seconds and test different things, but again this is a production system, so the quicker its back up, the better. =) Thanks, Marc -- _ -- 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] AST-2011-001: Stack buffer overflow in SIP channel driver
Could you please give me a feedback regarding this issue, I'm not sure of the
answer I got browsing the web
Thanks and Best Regards
Le mercredi 19 janvier 2011 09:14:55, Marc Leurent a écrit :
Good morning,
I have a simple question,
Is this problem would affect also an Asterisk 1.4.38 if Pedantic SIP
support: No in the Global Signalling Settings
For what I understood, no..
Or is it a simple way to postpone upgrade until next planned upgrade.
Best Regards
Le mardi 18 janvier 2011 17:35:31, Asterisk Security Team a écrit :
Asterisk Project Security Advisory - AST-2011-001
ProductAsterisk
SummaryStack buffer overflow in SIP channel driver
Nature of Advisory Exploitable Stack Buffer Overflow
SusceptibilityRemote Authenticated Sessions
Severity Moderate
Exploits KnownNo
Reported On January 11, 2011
Reported By Matthew Nicholson
Posted On January 18, 2011
Last Updated OnJanuary 18, 2011
Advisory Contact Matthew Nicholson mnichol...@digium.com
CVE Name
Description When forming an outgoing SIP request while in pedantic mode,
a
stack buffer can be made to overflow if supplied with
carefully crafted caller ID information. This vulnerability
also affects the URIENCODE dialplan function and in some
versions of asterisk, the AGI dialplan application as well.
The ast_uri_encode function does not properly respect the
size
of its output buffer and can write past the end of it when
encoding URIs.
Resolution The size of the output buffer passed to the ast_uri_encode
function is now properly respected.
In asterisk versions not containing the fix for this issue,
limiting strings originating from remote sources that will be
URI encoded to a length of 40 characters will protect against
this vulnerability.
exten = s,1,Set(CALLERID(num)=${CALLERID(num):0:40})
exten = s,n,Set(CALLERID(name)=${CALLERID(name):0:40})
exten = s,n,Dial(SIP/channel)
The CALLERID(num) and CALLERID(name) channel values, and any
strings passed to the URIENCODE dialplan function should be
limited in this manner.
Affected Versions
Product Release Series
Asterisk Open Source1.2.x All versions
Asterisk Open Source1.4.x All versions
Asterisk Open Source1.6.x All versions
Asterisk Open Source1.8.x All versions
Asterisk Business Edition C.x.x All versions
AsteriskNOW 1.5 All versions
s800i (Asterisk Appliance) 1.2.x All versions
Corrected In
Product Release
Asterisk Open Source 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1,
1.6.2.16.1, 1.8.1.2, 1.8.2.1
Asterisk Business Edition C.3.6.2
Patches
URL
Branch
http://downloads.asterisk.org/pub/security/AST-2011-001-1.4.diff1.4
http://downloads.asterisk.org/pub/security/AST-2011-001-1.6.1.diff
1.6.1
http://downloads.asterisk.org/pub/security/AST-2011-001-1.6.2.diff
1.6.2
Re: [asterisk-users] AST-2011-001: Stack buffer overflow in SIP channel driver
Thank you for the confirmation Best Regards, Le vendredi 21 janvier 2011 14:17:20, Kevin P. Fleming a écrit : On 01/21/2011 05:59 AM, Marc Leurent wrote: Could you please give me a feedback regarding this issue, I'm not sure of the answer I got browsing the web Thanks and Best Regards Le mercredi 19 janvier 2011 09:14:55, Marc Leurent a écrit : Good morning, I have a simple question, Is this problem would affect also an Asterisk 1.4.38 if Pedantic SIP support: No in the Global Signalling Settings For what I understood, no.. Or is it a simple way to postpone upgrade until next planned upgrade. The advisory clearly states that all Asterisk 1.4.x releases are affected when pedantic mode is enabled. Since you have pedantic mode disabled, your system is not vulnerable to 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] AST-2011-001: Stack buffer overflow in SIP channel driver
Good morning,
I have a simple question,
Is this problem would affect also an Asterisk 1.4.38 if Pedantic SIP
support: No in the Global Signalling Settings
For what I understood, no..
Or is it a simple way to postpone upgrade until next planned upgrade.
Best Regards
Le mardi 18 janvier 2011 17:35:31, Asterisk Security Team a écrit :
Asterisk Project Security Advisory - AST-2011-001
ProductAsterisk
SummaryStack buffer overflow in SIP channel driver
Nature of Advisory Exploitable Stack Buffer Overflow
SusceptibilityRemote Authenticated Sessions
Severity Moderate
Exploits KnownNo
Reported On January 11, 2011
Reported By Matthew Nicholson
Posted On January 18, 2011
Last Updated OnJanuary 18, 2011
Advisory Contact Matthew Nicholson mnichol...@digium.com
CVE Name
Description When forming an outgoing SIP request while in pedantic mode, a
stack buffer can be made to overflow if supplied with
carefully crafted caller ID information. This vulnerability
also affects the URIENCODE dialplan function and in some
versions of asterisk, the AGI dialplan application as well.
The ast_uri_encode function does not properly respect the size
of its output buffer and can write past the end of it when
encoding URIs.
Resolution The size of the output buffer passed to the ast_uri_encode
function is now properly respected.
In asterisk versions not containing the fix for this issue,
limiting strings originating from remote sources that will be
URI encoded to a length of 40 characters will protect against
this vulnerability.
exten = s,1,Set(CALLERID(num)=${CALLERID(num):0:40})
exten = s,n,Set(CALLERID(name)=${CALLERID(name):0:40})
exten = s,n,Dial(SIP/channel)
The CALLERID(num) and CALLERID(name) channel values, and any
strings passed to the URIENCODE dialplan function should be
limited in this manner.
Affected Versions
Product Release Series
Asterisk Open Source1.2.x All versions
Asterisk Open Source1.4.x All versions
Asterisk Open Source1.6.x All versions
Asterisk Open Source1.8.x All versions
Asterisk Business Edition C.x.x All versions
AsteriskNOW 1.5 All versions
s800i (Asterisk Appliance) 1.2.x All versions
Corrected In
Product Release
Asterisk Open Source 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1,
1.6.2.16.1, 1.8.1.2, 1.8.2.1
Asterisk Business Edition C.3.6.2
Patches
URL Branch
http://downloads.asterisk.org/pub/security/AST-2011-001-1.4.diff1.4
http://downloads.asterisk.org/pub/security/AST-2011-001-1.6.1.diff 1.6.1
http://downloads.asterisk.org/pub/security/AST-2011-001-1.6.2.diff 1.6.2
http://downloads.asterisk.org/pub/security/AST-2011-001-1.8.diff1.8
Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security
This document may be superseded by later versions; if so, the latest
[asterisk-users] Asterisk/Polycom Dialed Party Name
Hi, We are in the process of moving from an Avaya Definity to Asterisk for our institution's phone system. I got one feature that the Avaya had, which I have not been able to reproduce with Polycom phones and Asterisk; since this feature seemed so small and useless to me when testing, I kind of ignored it. Now I am getting more I miss that requests than I expected. =) On the Avaya's, when you dialed another user's internal extension, on the phone you are dialing from, it would display the user's name that you're dialing. Basically it was matching the contact/extension you dialed on your phone to an entry in the directory, and displaying that contact/extension's full name (eg, when I type in 20467 on my phone and hit dial, my phone would then display Marc Smith while it was ringing Marc Smith's phone). I've Google'd quite a bit and haven't really found any solutions; I found one other guy asking for the same thing, but no answer for him: http://forum.voxilla.com/polycom-voip-support-forum/see-you-call-polycom-ip330-37297.html Has anyone been able to replicate this with Asterisk and Polycom phones? If not with Polycoms, maybe another brand of phone? I see an option in the Polycom SIP administrator manual that comes close, but it clearly states this will not work for outgoing calls: --snip-- up.useDirectoryNames 0 or 1 Default: 0 If set to 1, the name fields of the local contact directory entries which match incoming calls will be used for caller identification display and in the call lists instead of the name provided through network signaling. Note: There is no matching of outgoing calls. There is no matching to corporate directory entries. --snip-- Currently on Asterisk 1.6.1.1 and Polycom SIP version 3.2.2.0477 (Polycom 330s and 560s). --Marc -- _ -- 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] Calculating R Factor and MOS metrics for VoIP
Take a look at http://dev.leurent.eu/voip/MOS/ I'v done this a long time ago, hope it will help! ++ Le 08.03.2010 11:10, mosbah.abdelkader a écrit : Hello All, MOS and R factor are the two QoS parameters used to estimate VoIP call quality. I have found that they are calculated from other metrics like jitter, latency, packet loss,...etc. But, haven't found any formula or arithmetic rule to calculate them. Do you have an idea about their formulas or an open source that calculates them. Is it possible to interpret them from wireshark. Thanks. -- Please discover scientific miracles of CORAN http://www.55a.net/ -- _ -- 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] Help getting info from caller
Can I use: exten = 33,n,Set(ACCOUNT=waitexten()) ??? No. Something like exten = 33,n,Read(ACCOUNT,,10) See http://www.voip-info.org/wiki/view/Asterisk+cmd+Read hth and happy new year to everyone ___ -- 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] Why no re-register when sip register status is UNREACHABLE
Hello, I had a situation with my asterisk server this morning. The DNS of the sip-server changed but Asterisk did not re-register to get the new IP adress. Is that normal? I do have the setting sip.conf registerattempts on default, meaning it should try unlimited number of times to register with the sip server. Before sip reload: sip-out/abc 172.16.200.1 N 5060 UNREACHABLE After reload: sip-out/abc 192.168.1.1 N 5060 OK (29 ms) Regards, Marc Ketel. ___ -- 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 1.4 DISA is jumping after one digit in the DISA context
Am Friday 06 November 2009 00:17:36 schrieb Marc Lindner: Dear list, I have problems with DISA on an specific server with Asterisk 1.4.26.2. After starting DISA I can only press one key and DISA is jumping direct into the context without waiting for further digits. The reason and solution is: exten = _X!,n,DISA(no-password|calls_disa) [calls_disa] exten = _X.,1,NoOp() exten = _X.,n,HangUp() if context [calls_disa] like this exten = _X!,1,NoOp() exten = _X!,n,HangUp() then DISA function is broken, after entering one digit, dialplan jump to calls_disa. I did not expected this... ___ -- 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 1.4 DISA is jumoing after one digit in the DISA context
Dear list, I have problems with DISA on an specific server with Asterisk 1.4.26.2. After starting DISA I can only press one key and DISA is jumping direct into the context without waiting for further digits. In dtmf.log I found this: [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin '7' received on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin passthrough '7' on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF end '7' received on SIP/214-00d92db0, duration 60 ms [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF end accepted with begin '7' on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF end '7' has duration 60 but want minimum 80, emulating on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF end emulation of '7' queued on SIP/214-00d92db0 If Iam using the dialplan on another server there is no problem. If Iam using READ I do not have problems to enter digits by DTMF so I assume its related to DISA. best regards Marc ___ -- 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] SIP Peers still ping with SIP OPTIONS on a reload
I have the same result with Asterisk 1.4.21 on a Debian Lenny server -- -- -- Marc LEURENT lf...@leurent.eu Le mercredi, 28 octobre 2009 12.27:59, Marc Leurent a écrit : Hello, when I remove a peer from my sip.conf and just do a reload, the peer is still ping with SIP OPTIONS until I restart Asterisk, I use Asterisk 1.4.27-rc2. Is it normal? Thanks As an example, I have added and after removed this lines and ;[sip_trk_vm] ;host=88.191.80.8 ;type=peer ;context=default ;dtmfmode=info ;insecure=port,invite ;nat=never ;sendrpid=yes ;disallow=all ;allow=alaw ___ -- 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] SIP Peers still ping with SIP OPTIONS on a reload
Hello, when I remove a peer from my sip.conf and just do a reload, the peer is still ping with SIP OPTIONS until I restart Asterisk, I use Asterisk 1.4.27-rc2. Is it normal? Thanks As an example, I have added and after removed this lines and ;[sip_trk_vm] ;host=88.191.80.8 ;type=peer ;context=default ;dtmfmode=info ;insecure=port,invite ;nat=never ;sendrpid=yes ;disallow=all ;allow=alaw -- -- -- Marc LEURENT lf...@leurent.eu ___ -- 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] How to generate 183 Session Progress
Thank you Klaus and Martin for your answers! It's very helpful! -- -- -- Marc LEURENT lf...@leurent.eu Le vendredi, 23 octobre 2009 20.51:54, Martin a écrit : You can call application Progress() from within dialplan and it will cause the Asterisk to send a SIP reply 183 on the call that came to Asterisk. Martin On Fri, Oct 23, 2009 at 6:36 AM, Marc Leurent lf...@leurent.eu wrote: Hello everybody, I have 2 users connected on the same Asterisk server that are connected with 2 different Asterisk servers. For outgoing calls, one in receiving 183 Session Progress and the other not! Do you have any idea why? Thanks. I have tried to understand by myself and in their INVITE they have almost the same Allow and Supported SIP Headers The one that works: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer The one that doen't work: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces -- -- -- Marc LEURENT lf...@leurent.eu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to generate 183 Session Progress
Hello everybody, I have 2 users connected on the same Asterisk server that are connected with 2 different Asterisk servers. For outgoing calls, one in receiving 183 Session Progress and the other not! Do you have any idea why? Thanks. I have tried to understand by myself and in their INVITE they have almost the same Allow and Supported SIP Headers The one that works: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer The one that doen't work: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces -- -- -- Marc LEURENT lf...@leurent.eu ___ -- 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] Dial a external number with extension
I need to make a conference between 2 numbers, one of them is external and it has an extension. So, I need to dial the number and later enter the extension, how can I do that? something like this : exten = 5145551212,Dial(Zap/g0/5145556000,20,D(7287)) see http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial hth ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No more room in scheduler
Hi, I running into the following problem on my Asterisk setup: --snip-- [Sep 3 01:40:59] NOTICE[9170] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 3 [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:09] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:09] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:09] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:09] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? --snip-- This happens once a week, at same about the same time (give or take a couple minutes). Always from span 3 too. It just continually spits out those messages until I restart Asterisk. I've seen others post about this, but haven't seen a real answer. Someone said to run a 'dahdi_test -v' when this happens; I did and I get 99% every time. Someone else said this is usually caused by the telco. running some type of test on the line, and I would agree since it happens every week at pretty much the same time and same day. So, yes, lets say the telco. is sending some type of signal that freaks out Asterisk/DAHDI. I could call them and ask them to stop, but it would seem more appropriate for Asterisk/DAHDI to just handle this and not cry. A short term fix would be to just have a cron run around 2:00 a.m. weekly that will restart Asterisk. Should I open a bug for this? asterisk-1.6.1.1 dahdi-linux-2.2.0.2 dahdi-tools-2.2.0 Linux jekyll.mcc.edu 2.6.18-128.1.1.el5 #1 SMP Mon Jan 26 13:58:24 EST 2009 x86_64 x86_64 x86_64 GNU/Linux Red Hat Enterprise Linux Server release 5.3 (Tikanga) Dell PowerEdge 2950 (2) Wildcard TE220 (4th Gen) [r...@jekyll ~]# cat /etc/dahdi/system.conf # 20090801 MAS # Span 1 span=1,1,0,esf,b8zs bchan=1-23 dchan=24 echocanceller=mg2,1-23 # Span 2 span=2,2,0,esf,b8zs bchan=25-47 dchan=48 echocanceller=mg2,25-47 # Span 3 span=3,3,0,esf,b8zs bchan=49-71 dchan=72 echocanceller=mg2,49-71 # Span 4 span=4,4,0,esf,b8zs bchan=73-95 dchan=96 echocanceller=mg2,73-95 # Global loadzone= us defaultzone = us [r...@jekyll ~]# cat /etc/asterisk/chan_dahdi.conf [general] [channels] ; Span 1 group = 1 context = from_pstn switchtype = qsig signalling = pri_net channel = 1-23 context = default ; Span 2 group = 2 context = from_avaya switchtype = qsig signalling = pri_net channel = 25-47 context = default ; Span 3 group = 7 context = from_pstn switchtype = qsig signalling = pri_cpe channel = 49-71 context = default ; Span 4 group = 7 context = from_pstn switchtype = qsig signalling = pri_cpe channel = 73-95 context = default [r...@jekyll ~]# cat /etc/dahdi/modules # 20090801 MAS wct4xxp wctc4xxp Let me know if any more information is needed. Any help is greatly appreciated! Thanks, Marc ___ -- 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] CLI file convert from alaw to g729 with TC400B transcoding card results to an empty file
Thank you Shaun for your answer! Indeed, I have made some basic tests to convert a file to g729 using the software codec and it works! Have a nice day! -- -- -- Marc LEURENT lf...@leurent.eu Le mercredi, 9 septembre 2009 19.32:30, Shaun Ruffell a écrit : On 09/09/2009 09:33 AM, Marc Leurent wrote: Good afternoon, I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card. The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty file as you can see below... CLI file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw /var/lib/asterisk/sounds/fr/service_notactivated.g729 (SCREEN):r...@nutella:[/var/../fr]# file service_notactivated* service_notactivated.alaw: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz service_notactivated.g729: empty service_notactivated.gsm: data I was able to create the gsm file with the command, but the g729 one is empty. Have you got any idea how I can solve this? Thanks PS: I'm able to place call in g729 without problem and the TC400B works well The problem is that 'convert' finishes when it receives back a NULL frame. For the software based codecs, this isn't a problem because the transcoding happens synchronously with the caller. However, codec_dahdi doesn't block the caller while the hardware is transcoding the audio, and therefore can return NULL frames if the hardware hasn't had enough time to transcode the frame. Probably what is needed is a standalone utility in dahdi-tools that can be used to transcode audio files with the hardware. Cheers, ___ -- 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] CLI file convert from alaw to g729 with TC400B transcoding card results to an empty file
Good afternoon, I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card. The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty file as you can see below... CLI file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw /var/lib/asterisk/sounds/fr/service_notactivated.g729 (SCREEN):r...@nutella:[/var/../fr]# file service_notactivated* service_notactivated.alaw: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz service_notactivated.g729: empty service_notactivated.gsm: data I was able to create the gsm file with the command, but the g729 one is empty. Have you got any idea how I can solve this? Thanks PS: I'm able to place call in g729 without problem and the TC400B works well -- -- -- Marc LEURENT lf...@leurent.eu ___ -- 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] Cisco 7960 Multiline phone
What I have is 1 front desk phone only with 6 lines Front Desk Phone line 1 - incoming extension 1 Front Desk Phone line 2 - incoming extension 2 Front Desk Phone line 3 - incoming extension 3 Front Desk Phone line 4 - incoming extension 4 Front Desk Phone line 5 - incoming extension 5 Front Desk Phone line 6 - inside office extension If incoming line 1 is busy I want the next incoming call to come in on line 2. If incoming line 2 and 3 are busy but 1 is free the next call should got to line 1. So lines 1 and 2 might get a lot of calls but only on really busy days will calls make it up to lines 4 and 5. Does that make sense? Anyone have the solution? You could probably use DEVICE_STATE to check the status first : http://www.voip-info.org/wiki/view/Asterisk+func+device_State hth ___ -- 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] Inquiry:Asterisk Inter digit delay
I mean when the subs dials the digits with some delay between entering the digits sequentially . At our current case , the Asterisk will wait about 2 seconds to see if another digit will be dialed or not and then he will route the dialed digits according to the pre-defined routing table or he will play the appropriate announcements . We are expected to increase this inter digit delay to say 4 seconds . Please let us know how we can increase this parameter in our Asterisk configuration files . see http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout also, I suggest you read this book : http://astbook.asteriskdocs.org/ hth ___ -- 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] Asterisk 1.4.26 final release - What is blocking?
Hello everybody, I was wondering what is postponing the 1.4.26 release? I thought it was scheculed for last week. Is there something we can do to help to release this version? There is no more issue reported on https://issues.asterisk.org/ for the time being. Best Regards, -- -- Marc LEURENT lf...@leurent.eu ___ -- 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] Sangoma A200 and bristuff how install
Hi, I want to install an Sangoma A200 together with an BRI card. I would like to use Asterisk 1.4 Are there any howto or tips? First compile bristuff and after compile wanpipe? 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] Open Source Call Statistics / Metrics Packages
Hi, Just wondering what the popular open source call statistics / metrics packages are for Asterisk? Preferably an all-in-one package that supports queues and calls from the CDR information generated by Asterisk. Whats everyone using? Favorites? Thanks, Marc ___ -- 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] MeetMe: Mute All Lines Automatically?
On Mon, Jun 8, 2009 at 9:18 AM, Christopher Stamperchristopherstam...@gmail.com wrote: I'm considering implementing an Asterisk PBX for conferencing. Before I get started, I wanted to make sure that it supports the features that I need. I plan to use Asterisk as a conference bridge only. I want people to be able to use my conference to listen live to lectures/etc, without having to listen to others in the conference. have a look at the documentation here : http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe you want this option 'm' — set monitor only mode (Listen only, no talking) hth ___ -- 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] Looking for good IAX ATA
On Fri, Apr 10, 2009 at 7:46 PM, John Rogers j...@wizworks.net wrote: Thank you for the links! Of course if anyone else knows of other IAX ATA offerings, please *DO* share. Really looking for a good solution. I will buy one of each of these offerings to test and I'll share my findings with the group. Didn't try them myself, but I found those 2 - http://x100p.com/products/FXS.php - http://www.atcom.cn/En_products_AG188N.html hth ___ -- 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] Extract a MOS value from Asterisk CDR
Indeed, we already have
- the function to convert R factor to MOS
- the R function R = R0 -Is-Id-Ie+A
- the codec used
- the rtt, rx/tx jitter, packet loss
What ye do not have but is needed:
- A factor, a note between 0 and 20 - 0 for landlines
- the Burst Ratio, I'm using 1 (random repartition)
I already have an openoffice calc function to calculate the MOS regarding the
rtt, packet loss, codec, I have to add the jitter!
Here are the URL I have used
* http://www.itu.int/rec/T-REC-G.107-200503-S/en
* http://www.ixiacom.com/library/white_papers/display?skey=voip_quality
* http://www.itu.int/ITU-T/studygroups/com12/emodelv1/tut.htm
Have a nice day!
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch - marc.leur...@vtx-telecom.ch
VTX, votre partenaire telecom proche de vous !
Le Thursday 02 April 2009 03.27:48 John Todd, vous avez écrit :
Thank you for the interesting links on MOS values and calculations!
It seems that many (most?) of the values that are used to construct R
and MOS could be obtained from the data that exists within the
dialplan, at least as far as the visible RTP path is concerned. Or
is there data missing in the current RTCP statistics that would be
required to make correct R/MOS value estimates? (If so, then that's
on-topic for asterisk-dev, otherwise this should be moved to asterisk-
users...)
Here is the data that I think is already visible:
- codec choices
- round-trip delay to RTP endpoint
- packet loss
- jitter
I think it is too complex to determine Irecency, A or packet loss
bursts unless there is significant additional code added to Asterisk
to capture more granular time-slices of data on each call. I also
think that mid-call codec changes should not be considered due to
complexity. Currently, I think this is un-necessary since most people
don't even seem to compute MOS to start with.
So in your examination you may come up with a script or dialplan that
creates a synthetic R or MOS value - could you post it to a blog, or
if it is very short, to the asterisk-users mailing list? I think this
would be worthwhile.
JT
On Apr 1, 2009, at 2:57 PM, Mindaugas Kezys wrote:
Sorry for replying for the second time, but this issue is
interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this:
http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc
Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from
Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to
calculate a MOS
value:
ssrc
=
592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
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---
John Todd email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083 http://www.digium.com/
___
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--
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch
[asterisk-users] [CLOSED] Asterisk + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk
Hello, all.
This is just an email to inform you I have added a SIP header in Asterisk SIP
message that is handled by the proxy:
On Asterisk extensions.conf:
SIPAddHeader(X-number-to-dial: ${NUMBERTOREACH})
Dial(SIP/${MAINPEER}|100|t)
and on OpenSIPS:
if (is_present_hf(X-number-to-dial)) {
xlog(L_DBG, GOING TO replace URI username with X-number-to-dial\r\n);
xlog(L_DBG, Print $(hdr(X-number-to-dial)) \r\n);
subst_user('/(.*)/$(hdr(X-number-to-dial))/');# Substitute the URI
phone number with the one in X-number-to-dial SIP Header
subst('/^(To|t):(.*)sip:[...@]*@(.*)$/\1:\2sip:
$(hdr(X-number-to-dial))@\3/ig');
}
Have a nice day!
-- --
Marc LEURENT
Le Monday 23 March 2009 13.41:59 Marc Leurent, vous avez écrit :
I have spoken to quickly,
Usually Asterisk on an incoming call sends an INVITE
Reg.Contact
Number@Reg Contact IP to the Peer IP. With the command you gave me, it
is possible to send anINVITE othernumber@Peer IP to the
Peer IP.
What I would like to do is to sendINVITE
othernumber@Reg Contact IP
to the Peer IP in order for the request to be forwarded by the proxy!
Is it possible to do something like:
Dial(SIP/sip:1...@192.168.10.125:5060@1003 )
in Order to send INVITE 1...@1005 IP to 1003 device IP
Thanks!
Le Monday 23 March 2009 12.03:55 Marc Leurent, vous avez écrit :
Thank you, this is exactly what I needed!!
In order to Dial any number to a registered peer, I just have to enter
Dial(SIP/anynum...@sippeername) Best Regards!
Le Monday 23 March 2009 11.31:31 Alex Balashov, vous avez écrit :
The Request URI generated in an INVITE originated by Asterisk is
governed entirely by the parameters passed to Dial().
For example:
Dial(SIP/1...@peer_name)
... will generate a Request URI of
1...@host.or.ip.of.sip.conf.peer.named.peer_name.
It is also possible to send requests to hosts that are not explicitly
defined in sip.conf, with the caveat that only background [general]
sip.conf settings will then apply:
Dial(SIP/1...@ip.of.peer.not.in.sip.conf)
Marc Leurent wrote:
Hello,
it is not an OpenSIPs problem I have, it's an Asterisk one,
I would like to change the URI in message generated by Asterisk.
Thanks
Le Monday 23 March 2009 10.35:09 Alex Balashov, vous avez écrit :
Modify the $ru pseudovariable or use rewritehostport() out of core.
This is not the right mailing list. This belongs on the
OpenSIPS/OpenSER lists.
There is also a mailing list we operate called
SER-Asterisk-Interwork that is specifically intended to address SER*
/ Asterisk integration issues:
http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork
* Anything from the [Open]SER family.
lftsy wrote:
Hye everybody, anyone has any idea how to help me?
To resume, I just want to know how to change the IP in the URI sent
by Asterisk (first line of SIP packets)
Thanks for your time!
++
On Fri, 20 Mar 2009 15:09:55 +0100, Marc Leurent lf...@leurent.eu
wrote:
Hello All,
I have a little complicated question about the Dial command.
I use OpenSIPs to loadbalance Asterisk Servers, and Users are
registered on Asterisk servers.
Asterisk use the Reg. Contact entry to reach the UAC via the
OpenSIPs server. Everything works except for trunk numbers:
For each peer on Asterisk, Addr-IP is IP of the Proxy and Reg.
Contact is the IP where the proxy will relay the packet to reach
the
UAC.
Ex: with a trunk 0123400010 - 0123400019 with 0123400010 as the
sip
peer.
When a number from a trunk is called, like 0123400019 the Reg.
Contact of the main number is not used.
For the time being, I use Dial(SIP/0123400010/0123400019) but it
It sends an
INVITE sip:0123400...@proxyip to the proxy
whereas it should send
INVITE sip:0123400019@Reg. Contact of the main number to the
proxy
So I'm trying use the Dial Command with
Dial(SIP/0123400010/0123400019@Reg. Contact of the main number)
but it doesn't work
Have you got any idea how to rewrite the IP of the URI sent?
Thanks!
--
-- --
Marc LEURENT
lf...@leurent.eu
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--
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--
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--
-- --
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lf...@leurent.eu
Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all, I have put my MOS.ods file into
http://dev.leurent.eu/voip/MOS/
My problem is to add the jitter value into the formula
Have you got any idea how to do it?
-- --
Marc LEURENT
Le Thursday 02 April 2009 11.20:06 Mindaugas Kezys, vous avez écrit :
Could you share with us your Openoffice callc function?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 2 d. 11:29
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Indeed, we already have
- the function to convert R factor to MOS
- the R function R = R0 -Is-Id-Ie+A
- the codec used
- the rtt, rx/tx jitter, packet loss
What ye do not have but is needed:
- A factor, a note between 0 and 20 - 0 for landlines
- the Burst Ratio, I'm using 1 (random repartition)
I already have an openoffice calc function to calculate the MOS regarding the
rtt, packet loss, codec, I have to add the jitter!
Here are the URL I have used
* http://www.itu.int/rec/T-REC-G.107-200503-S/en
* http://www.ixiacom.com/library/white_papers/display?skey=voip_quality
* http://www.itu.int/ITU-T/studygroups/com12/emodelv1/tut.htm
Have a nice day!
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch - marc.leur...@vtx-telecom.ch
VTX, votre partenaire telecom proche de vous !
Le Thursday 02 April 2009 03.27:48 John Todd, vous avez écrit :
Thank you for the interesting links on MOS values and calculations!
It seems that many (most?) of the values that are used to construct R
and MOS could be obtained from the data that exists within the
dialplan, at least as far as the visible RTP path is concerned. Or
is there data missing in the current RTCP statistics that would be
required to make correct R/MOS value estimates? (If so, then that's
on-topic for asterisk-dev, otherwise this should be moved to asterisk-
users...)
Here is the data that I think is already visible:
- codec choices
- round-trip delay to RTP endpoint
- packet loss
- jitter
I think it is too complex to determine Irecency, A or packet loss
bursts unless there is significant additional code added to Asterisk
to capture more granular time-slices of data on each call. I also
think that mid-call codec changes should not be considered due to
complexity. Currently, I think this is un-necessary since most people
don't even seem to compute MOS to start with.
So in your examination you may come up with a script or dialplan that
creates a synthetic R or MOS value - could you post it to a blog, or
if it is very short, to the asterisk-users mailing list? I think this
would be worthwhile.
JT
On Apr 1, 2009, at 2:57 PM, Mindaugas Kezys wrote:
Sorry for replying for the second time, but this issue is
interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this:
http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc
Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from
Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to
calculate a MOS
value:
ssrc
=
592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
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Re: [asterisk-users] FXO Ignore ring
On Thu, Apr 2, 2009 at 11:37 AM, Cary Fitch ca...@usawide.net wrote: Is there a way to program an FXO device to totally ignore incoming calls? put the port in that context : [incoming-noanswer] exten = s,1,Hangup() hth ___ -- 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] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to calculate a MOS
value:
ssrc=592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
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Re: [asterisk-users] Asterisk + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk
Hello, it is not an OpenSIPs problem I have, it's an Asterisk one, I would like to change the URI in message generated by Asterisk. Thanks Le Monday 23 March 2009 10.35:09 Alex Balashov, vous avez écrit : Modify the $ru pseudovariable or use rewritehostport() out of core. This is not the right mailing list. This belongs on the OpenSIPS/OpenSER lists. There is also a mailing list we operate called SER-Asterisk-Interwork that is specifically intended to address SER* / Asterisk integration issues: http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork * Anything from the [Open]SER family. lftsy wrote: Hye everybody, anyone has any idea how to help me? To resume, I just want to know how to change the IP in the URI sent by Asterisk (first line of SIP packets) Thanks for your time! ++ On Fri, 20 Mar 2009 15:09:55 +0100, Marc Leurent lf...@leurent.eu wrote: Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, Addr-IP is IP of the Proxy and Reg. Contact is the IP where the proxy will relay the packet to reach the UAC. Ex: with a trunk 0123400010 - 0123400019 with 0123400010 as the sip peer. When a number from a trunk is called, like 0123400019 the Reg. Contact of the main number is not used. For the time being, I use Dial(SIP/0123400010/0123400019) but it It sends an INVITE sip:0123400...@proxyip to the proxy whereas it should send INVITE sip:0123400019@Reg. Contact of the main number to the proxy So I'm trying use the Dial Command with Dial(SIP/0123400010/0123400019@Reg. Contact of the main number) but it doesn't work Have you got any idea how to rewrite the IP of the URI sent? Thanks! -- -- -- Marc LEURENT lf...@leurent.eu ___ -- 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 -- -- -- Marc LEURENT lf...@leurent.eu ___ -- 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 + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk
Thank you, this is exactly what I needed!! In order to Dial any number to a registered peer, I just have to enter Dial(SIP/anynum...@sippeername) Best Regards! Le Monday 23 March 2009 11.31:31 Alex Balashov, vous avez écrit : The Request URI generated in an INVITE originated by Asterisk is governed entirely by the parameters passed to Dial(). For example: Dial(SIP/1...@peer_name) ... will generate a Request URI of 1...@host.or.ip.of.sip.conf.peer.named.peer_name. It is also possible to send requests to hosts that are not explicitly defined in sip.conf, with the caveat that only background [general] sip.conf settings will then apply: Dial(SIP/1...@ip.of.peer.not.in.sip.conf) Marc Leurent wrote: Hello, it is not an OpenSIPs problem I have, it's an Asterisk one, I would like to change the URI in message generated by Asterisk. Thanks Le Monday 23 March 2009 10.35:09 Alex Balashov, vous avez écrit : Modify the $ru pseudovariable or use rewritehostport() out of core. This is not the right mailing list. This belongs on the OpenSIPS/OpenSER lists. There is also a mailing list we operate called SER-Asterisk-Interwork that is specifically intended to address SER* / Asterisk integration issues: http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork * Anything from the [Open]SER family. lftsy wrote: Hye everybody, anyone has any idea how to help me? To resume, I just want to know how to change the IP in the URI sent by Asterisk (first line of SIP packets) Thanks for your time! ++ On Fri, 20 Mar 2009 15:09:55 +0100, Marc Leurent lf...@leurent.eu wrote: Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, Addr-IP is IP of the Proxy and Reg. Contact is the IP where the proxy will relay the packet to reach the UAC. Ex: with a trunk 0123400010 - 0123400019 with 0123400010 as the sip peer. When a number from a trunk is called, like 0123400019 the Reg. Contact of the main number is not used. For the time being, I use Dial(SIP/0123400010/0123400019) but it It sends an INVITE sip:0123400...@proxyip to the proxy whereas it should send INVITE sip:0123400019@Reg. Contact of the main number to the proxy So I'm trying use the Dial Command with Dial(SIP/0123400010/0123400019@Reg. Contact of the main number) but it doesn't work Have you got any idea how to rewrite the IP of the URI sent? Thanks! -- -- -- Marc LEURENT lf...@leurent.eu ___ -- 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 -- -- -- Marc LEURENT lf...@leurent.eu ___ -- 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 + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk
I have spoken to quickly, Usually Asterisk on an incoming call sends an INVITE Reg.Contact Number@Reg Contact IP to the Peer IP. With the command you gave me, it is possible to send an INVITE othernumber@Peer IP to the Peer IP. What I would like to do is to send INVITE othernumber@Reg Contact IP to the Peer IP in order for the request to be forwarded by the proxy! Is it possible to do something like: Dial(SIP/sip:1...@192.168.10.125:5060@1003 ) in Order to send INVITE 1...@1005 IP to 1003 device IP Thanks! Le Monday 23 March 2009 12.03:55 Marc Leurent, vous avez écrit : Thank you, this is exactly what I needed!! In order to Dial any number to a registered peer, I just have to enter Dial(SIP/anynum...@sippeername) Best Regards! Le Monday 23 March 2009 11.31:31 Alex Balashov, vous avez écrit : The Request URI generated in an INVITE originated by Asterisk is governed entirely by the parameters passed to Dial(). For example: Dial(SIP/1...@peer_name) ... will generate a Request URI of 1...@host.or.ip.of.sip.conf.peer.named.peer_name. It is also possible to send requests to hosts that are not explicitly defined in sip.conf, with the caveat that only background [general] sip.conf settings will then apply: Dial(SIP/1...@ip.of.peer.not.in.sip.conf) Marc Leurent wrote: Hello, it is not an OpenSIPs problem I have, it's an Asterisk one, I would like to change the URI in message generated by Asterisk. Thanks Le Monday 23 March 2009 10.35:09 Alex Balashov, vous avez écrit : Modify the $ru pseudovariable or use rewritehostport() out of core. This is not the right mailing list. This belongs on the OpenSIPS/OpenSER lists. There is also a mailing list we operate called SER-Asterisk-Interwork that is specifically intended to address SER* / Asterisk integration issues: http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork * Anything from the [Open]SER family. lftsy wrote: Hye everybody, anyone has any idea how to help me? To resume, I just want to know how to change the IP in the URI sent by Asterisk (first line of SIP packets) Thanks for your time! ++ On Fri, 20 Mar 2009 15:09:55 +0100, Marc Leurent lf...@leurent.eu wrote: Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, Addr-IP is IP of the Proxy and Reg. Contact is the IP where the proxy will relay the packet to reach the UAC. Ex: with a trunk 0123400010 - 0123400019 with 0123400010 as the sip peer. When a number from a trunk is called, like 0123400019 the Reg. Contact of the main number is not used. For the time being, I use Dial(SIP/0123400010/0123400019) but it It sends an INVITE sip:0123400...@proxyip to the proxy whereas it should send INVITE sip:0123400019@Reg. Contact of the main number to the proxy So I'm trying use the Dial Command with Dial(SIP/0123400010/0123400019@Reg. Contact of the main number) but it doesn't work Have you got any idea how to rewrite the IP of the URI sent? Thanks! -- -- -- Marc LEURENT lf...@leurent.eu ___ -- 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 -- -- -- Marc LEURENT lf...@leurent.eu ___ -- 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 + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk
Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, Addr-IP is IP of the Proxy and Reg. Contact is the IP where the proxy will relay the packet to reach the UAC. Ex: with a trunk 0123400010 - 0123400019 with 0123400010 as the sip peer. When a number from a trunk is called, like 0123400019 the Reg. Contact of the main number is not used. For the time being, I use Dial(SIP/0123400010/0123400019) but it It sends an INVITE sip:0123400...@proxyip to the proxy whereas it should send INVITE sip:0123400019@Reg. Contact of the main number to the proxy So I'm trying use the Dial Command with Dial(SIP/0123400010/0123400019@Reg. Contact of the main number) but it doesn't work Have you got any idea how to rewrite the IP of the URI sent? Thanks! -- -- -- Marc LEURENT lf...@leurent.eu ___ -- 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] OpenSIPS on CentOS
Hello Darrin, Maybe you should ask this question on OpenSIPs mailing list. I have build a rpm for CentOS 5.2 using and updated opensips.spec from svn 1) retrieve opensips.init and opensips.spec-4.4 from https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.4/packaging/rpm 2) retrieve http://www.opensips.org/pub/opensips/1.4.4/src/opensips-1.4.4-tls_src.tar.gz 3) put opensips.init and opensips-1.4.4-tls_src.tar.gz in /usr/src/redhat/SOURCES 4) put opensips.spec-4.4 in /usr/src/redhat/SPECS 5) run rpmbuild -bb opensips.spec-4.4 (and install missing build dependencies if necessary) ++ Le Friday 20 March 2009 15.19:07 Darrin Henshaw, vous avez écrit : I’ve been looking into OpenSIPS to see if it’s a worthwhile addition to our setup. We’re currently running a cluster, using Heartbeat, between two servers. It works well but I’m interested in seeing if we can improve it. My manager heavily uses RPM’s for installations rather than source, particularly using yum to update. I’m trying to actually install OpenSips via that method. Does anyone have/know of where I can find an rpm aimed at EL4? And how I can nab libxml2, with the dependencies. I can get an RPM from for libxml2 from ftp://xmlsoft.org/libxml2/, but the dependencies for it are causing me headaches. Any suggestions would be helpful. Thanks. -- -- -- Marc LEURENT Ingénieur VoIP DECKPOINT SA Une société du groupe VTX Telecom Rue Eugène-Marziano 15 - 1227 Les Acacias http://www.vtx.ch - marc.leur...@vtx-telecom.ch VTX, votre partenaire telecom proche de vous ! ___ -- 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] [OpenSIPS-Users] OpenSIPS on CentOS
Hello Bogdan, I have set a small rpm repository for opensips 4.4 for CentOS (with el5) x86_64 (and later i386 32bits) Simply visit http://centos.leurent.eu/ and read the README.txt Maybe we could just do the same on the official OpenSIPs website? ++ Le Friday 20 March 2009 18.12:25 Bogdan-Andrei Iancu, vous avez écrit : Hi Darrin, Hi Marc, Darrin, with an OpenSIPS frontend you can do more things actually: 1) move the HA in OpenSIPS - it will be able to re-route if one of the Asterisk boxs is down 2) do LB - you can use in parallel multiple Asterisk boxes and to balance the traffic between 3) you can terminate TLS (from client) and convert to UDP to deliver to Asterisk. Marc, Darrin has a point here - if you want to give a quick try to something, it is nice to be able to install it easily. We already have an APT (for debian) repo up and running (still beta). We could do the same for RPMs or, in the worst case, to generate the packages for download. Also, there are some RPMs (for suse) - see http://www.opensips.org/index.php?n=Resources.Downloads Regards, Bogdan Marc Leurent wrote: Hello Darrin, Maybe you should ask this question on OpenSIPs mailing list. I have build a rpm for CentOS 5.2 using and updated opensips.spec from svn 1) retrieve opensips.init and opensips.spec-4.4 from https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.4/packaging/rpm 2) retrieve http://www.opensips.org/pub/opensips/1.4.4/src/opensips-1.4.4-tls_src.tar.gz 3) put opensips.init and opensips-1.4.4-tls_src.tar.gz in /usr/src/redhat/SOURCES 4) put opensips.spec-4.4 in /usr/src/redhat/SPECS 5) run rpmbuild -bb opensips.spec-4.4 (and install missing build dependencies if necessary) ++ Le Friday 20 March 2009 15.19:07 Darrin Henshaw, vous avez écrit : I’ve been looking into OpenSIPS to see if it’s a worthwhile addition to our setup. We’re currently running a cluster, using Heartbeat, between two servers. It works well but I’m interested in seeing if we can improve it. My manager heavily uses RPM’s for installations rather than source, particularly using yum to update. I’m trying to actually install OpenSips via that method. Does anyone have/know of where I can find an rpm aimed at EL4? And how I can nab libxml2, with the dependencies. I can get an RPM from for libxml2 from ftp://xmlsoft.org/libxml2/, but the dependencies for it are causing me headaches. Any suggestions would be helpful. Thanks. -- -- -- Marc LEURENT Ingénieur VoIP DECKPOINT SA Une société du groupe VTX Telecom Rue Eugène-Marziano 15 - 1227 Les Acacias http://www.vtx.ch - marc.leur...@vtx-telecom.ch VTX, votre partenaire telecom proche de vous ! ___ -- 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] Good phone near $125
I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) I like the Polycom IP-330. 2 lines, nice speakerphone, dual ethernet, support PoE and works with 2.5mm headset. $110 at voipsupply ___ -- 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] Pingable and Unreachable at the same time !
Asterisk doesn't use PING to check the STATUS, it uses a SIP OPTION message. Regards, Marc From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: mardi 17 février 2009 14:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Pingable and Unreachable at the same time ! Hi, Has anyone met something like this ? dialor*CLI sip show peers Name/username HostDyn Nat ACL Port Status 7541/7541 (Unspecified)D 0UNKNOWN 7540/7540 (Unspecified)D 0UNKNOWN 7534/7534 (Unspecified)D 0UNKNOWN 7533/7533 (Unspecified)D 0UNKNOWN 7531/7531 192.168.100.199 D 5060 OK (10 ms) 7530/7530 192.168.100.196 D 5060 UNREACHABLE patton/patton 192.168.100.52 D 5060 OK (33 ms) trunk/trunk4ipbx 192.168.64.25060 OK (1 ms) 8 sip peers [Monitored: 3 online, 5 offline Unmonitored: 0 online, 0 offline] dialor*CLI !ping 192.168.100.196 PING 192.168.100.196 (192.168.100.196) 56(84) bytes of data. 64 bytes from 192.168.100.196http://192.168.100.196: icmp_seq=1 ttl=64 time=0.334 ms 64 bytes from 192.168.100.196http://192.168.100.196: icmp_seq=2 ttl=64 time=0.305 ms 64 bytes from 192.168.100.196http://192.168.100.196: icmp_seq=3 ttl=64 time=0.305 ms Any explaination ? Regards ___ -- 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] Faxing with asterisk
The Attrafax software that was mentioned at the beginning of the thread does support Gateway mode. Regards, Marc -Original Message- Fabio Mosti wrote: 2009/2/16 Steve Underwood ste...@coppice.org: You don't indicate the kind of setup you are using. I use asterisk (Spandsp) with a IAX2 trunk (ethernet connection) to another asterisk (zap). client-asterisk (Spandsp)-asterisk (zap)-fax To quote the Mythbusters, there's your problem. Fax over IP = forget it unless the connection between your two Asterisk machines is some form of LAN connection. This *may* change a little when the T.38 support in Asterisk includes a gateway mode, which I don't believe it does yet. (IIRC 1.6 includes much better support for T.38, but I don't think it includes this kind of gateway yet - anyone care to correct me?) ___ -- 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] Devstate and Voicemail
Have an interesting problem, Using asterisk 1.6.0.1 Phone A receives voicemail, dials into VoiceMailMain, Phone B's BLF for A lights up. Phone A deletes the voicemail but still in VoiceMailMain, Phone B's BLF for A goes off. Phone A hang's up, Phone B's BLF for A goes on. From this point forward the Phone B's BLF for A seems to always show the opposite of what it should. I've looked at 'core show hints' and it is in fact reporting INUSE when it's not, and NOT_INUSE when it is. Is this a bug or just some configuration option that I am missing? Is there a way to manually change the devicestate for a channel? Marc Hudson ___ -- 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] Devstate and Voicemail
Philipp Kempgen wrote: Jared Smith schrieb: On Thu, 2008-10-23 at 09:36 -0700, Marc Hudson wrote: I've looked at 'core show hints' and it is in fact reporting INUSE when it's not, and NOT_INUSE when it is. That definitely sounds like a bug to me. Could you please report this on the bug tracker, so that the developers can take a look and try to reproduce and solve the problem? Sounds a bit like http://bugs.digium.com/view.php?id=13668 or http://bugs.digium.com/view.php?id=13238 Maybe they're all related to each other. Philipp Kempgen Yeah, looks like http://bugs.digium.com/view.php?id=13668, getting -1/0/0 in 'sip show inuse'. Odd that VoiceMailMain of all things happened to trigger it in this case. Thanks, Marc Hudson ___ -- 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 AGI and php problem....
'/var/lib/asterisk/agi-bin/cid-to-acct.php': No such file or directory 2 == cid-to-acct.php: Failed to execute It is not complaining about the lack of /usr/bin/php, but about the fact that the file /var/lib/asterisk/agi-bin/cid-to-acct.php is nowhere to be found. Probably asking the obvious but... Did you place the file in the agi-bin folder ? Is it really named cid-to-acct.php ? Is it executable ? Does the user under which asterisk is running as the right to execute it ? hth ___ -- 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] sip prune realtime per issue
On Tue, Jul 15, 2008 at 12:05 PM, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: I am using realtime on two boxes, one running 1.4.10.1 and one running 1.4.11. Everything works fine except for when I make a database change, such as a phones password. I change the DB, I prune the peer, I see it is gone and then I see it show up again in sip show peer , but everything is not being updated. The phone will not register even though the DB and the phone have the correct password. The only way to get it to register is to stop * and re-start it, then it works fine. I even tried changing the callerid and pruned the peer. A sip show peer shows the correct callerid, but when you call into voicemail, it is using the old callerid. Again, if I stop * and restart, it works fine. Has anybody seen this bug and if so, know what the bug ID is? We have a bunch of patches on these boxes and can't just upgrade to any old version to see if it fixes it. I need to figure out what the bug is. I did some research, but couldn't find it. Peder Do the rt* options in sip.conf have any effect? Maybe one of those might help? --Marc ___ -- 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 !DSPAM:1,487ccb5365666785646901! ___ -- 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] IMAP Storage Problem
On Fri, Jul 11, 2008 at 11:46 PM, Marc Smith [EMAIL PROTECTED] wrote:
Hi,
I'm having a problem with IMAP storage and asterisk. Here is the error
message I get (in this instance its checking messages):
[Jul 11 23:14:12] WARNING[9888]: app_voicemail.c:8738 mm_log: IMAP
Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN
[Jul 11 23:14:12] ERROR[9888]: app_voicemail.c:8741 mm_log: IMAP
Error: IMAP protocol error: Authentication aborted
[Jul 11 23:14:12] ERROR[9888]: app_voicemail.c:8741 mm_log: IMAP
Error: IMAP Authentication cancelled
[Jul 11 23:14:12] ERROR[9888]: app_voicemail.c:4790 init_mailstream:
Can't connect to imap server
{mail.host.com:143/imap/notls/user=bigtizzies}INBOX
[Jul 11 23:14:12] ERROR[9888]: app_voicemail.c:2486 messagecount: IMAP
mailstream is NULL
voicemail.conf:
[general]
imapserver=mail.host.com
imapport=143
imapflags=notls
[default]
20002 = 1234,Sue's Mailbox,,,imapuser=bigtizzies|imapsecret=largedillas
Yet, when doing a 'mtest' (from the uw-imap directory I used for
asterisk) with {mail.host.com:143/imap/notls/user=bigtizzies}INBOX
and it works fine.
I seen a post on the Digium forums
(http://forums.digium.com/viewtopic.php?t=14432highlight=imap) where
another person had this same problem and he said he fixed it by fixing
a typo -- I've looked over my config and all seems good.
I'm attempting to connect to dovecot, here is a snip of the log on the
IMAP server:
Jul 11 23:26:04 esdiaz dovecot: imap-login: Aborted login (1
authentication attempts): method=PLAIN, rip=10.100.100.100,
lip=207.73.29.38
Anyone else ran across something like this? Ideas?
Thanks,
Marc
Update:
It appears imapsecret is incorrect; when I change that field to
imappassword it works perfectly!
--Marc
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Re: [asterisk-users] Google Apps IMAP
}asterisk-users
[Jul 14 14:28:43] DEBUG[8837] app_voicemail.c: Before mail_open,
server: {imap.gmail.com:993/imap/ssl/[EMAIL PROTECTED],
box:0
[Jul 14 14:28:43] DEBUG[8837] app_voicemail.c: IMAP Info: Reusing
connection to gmail-imap.l.google.com/user=[EMAIL PROTECTED]
[Jul 14 14:28:43] DEBUG[8837] app_voicemail.c: Entering EXISTS
callback for message 2397
[Jul 14 14:28:43] DEBUG[8837] app_voicemail.c: User
[EMAIL PROTECTED] mailbox set for update.
[Jul 14 14:28:43] DEBUG[8837] app_voicemail.c: IMAP Info: [UNSEEN 2]
[Jul 14 14:28:43] DEBUG[8837] res_config_mysql.c: MySQL RealTime:
Retrieve SQL: SELECT * FROM voicemail WHERE mailbox = '20001' AND
context
= 'default'
[Jul 14 14:28:43] DEBUG[8837] app_voicemail.c:
[EMAIL PROTECTED] not found in vmstates
[Jul 14 14:28:43] DEBUG[8837] app_voicemail.c: Mailbox set to 20001
[Jul 14 14:28:43] DEBUG[8837] app_voicemail.c: comparing mailbox
20001 (i=1) to vmstate mailbox 20001 (i=0)
[Jul 14 14:28:43] DEBUG[8837] app_voicemail.c: 20001 not found in vmstates
[Jul 14 14:28:43] DEBUG[8837] app_voicemail.c: vm_state user
is:[EMAIL PROTECTED]
[Jul 14 14:28:43] DEBUG[8837] app_voicemail.c: Before mail_open,
server: {imap.gmail.com:993/imap/ssl/[EMAIL PROTECTED],
box:1
UNSEEN 2 -- So, it sees I have 2 unread messages.
I'm starting to doubt if GMail/Asterisk-IMAP integration is going to
be right for our institution, even if I get this working correctly,
there is still the problem of how to turn on IMAP access for all GApps
accounts and make it stay on. I don't believe the GApps API supports
changing user's options that way.
--Marc
2008/6/25 Marc Smith [EMAIL PROTECTED]:
Hi,
Anyone using Asterisk IMAP voicemail storage with Google Apps / GMail
IMAP? If so, does their IMAP implementation support any kind of
master user (Dovecot) abililty? Good? Bad?
--Marc
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http://www.suretecsystems.com/services/openldap/
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!DSPAM:1,4862c35665662617731437!
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[asterisk-users] IMAP Storage Problem
Hi,
I'm having a problem with IMAP storage and asterisk. Here is the error
message I get (in this instance its checking messages):
[Jul 11 23:14:12] WARNING[9888]: app_voicemail.c:8738 mm_log: IMAP
Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN
[Jul 11 23:14:12] ERROR[9888]: app_voicemail.c:8741 mm_log: IMAP
Error: IMAP protocol error: Authentication aborted
[Jul 11 23:14:12] ERROR[9888]: app_voicemail.c:8741 mm_log: IMAP
Error: IMAP Authentication cancelled
[Jul 11 23:14:12] ERROR[9888]: app_voicemail.c:4790 init_mailstream:
Can't connect to imap server
{mail.host.com:143/imap/notls/user=bigtizzies}INBOX
[Jul 11 23:14:12] ERROR[9888]: app_voicemail.c:2486 messagecount: IMAP
mailstream is NULL
voicemail.conf:
[general]
imapserver=mail.host.com
imapport=143
imapflags=notls
[default]
20002 = 1234,Sue's Mailbox,,,imapuser=bigtizzies|imapsecret=largedillas
Yet, when doing a 'mtest' (from the uw-imap directory I used for
asterisk) with {mail.host.com:143/imap/notls/user=bigtizzies}INBOX
and it works fine.
I seen a post on the Digium forums
(http://forums.digium.com/viewtopic.php?t=14432highlight=imap) where
another person had this same problem and he said he fixed it by fixing
a typo -- I've looked over my config and all seems good.
I'm attempting to connect to dovecot, here is a snip of the log on the
IMAP server:
Jul 11 23:26:04 esdiaz dovecot: imap-login: Aborted login (1
authentication attempts): method=PLAIN, rip=10.100.100.100,
lip=207.73.29.38
Anyone else ran across something like this? Ideas?
Thanks,
Marc
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[asterisk-users] Google Apps IMAP
Hi, Anyone using Asterisk IMAP voicemail storage with Google Apps / GMail IMAP? If so, does their IMAP implementation support any kind of master user (Dovecot) abililty? Good? Bad? --Marc ___ -- 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] connecting 2 FXS together
On Wed, Jun 4, 2008 at 9:04 AM, Steven Howes [EMAIL PROTECTED] wrote: On 4 Jun 2008, at 11:43, Joao Ferreira gmail wrote: can I connect 2 FXS plugs to the same analog phone ? No. Fire and death. Unless you use a 2-lines analog phone :) ___ -- 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] voicemail custom greeting
I have a wav file recording that i want to use on my voicemail, how can i set this up? You could play that file before sending the person to your voicemail and pass the s option to it Type show application voicemail on asterisk CLI to see the options. hth ___ -- 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] what's a softphone can activer web browser
can anyone help me. I'm finding the softphone which can trigger web browser and use callerid to go web page You don't say on what OS you need it to run. Mine is for Windows and support receiving URL (ex.: Dial(IAX2/7003|20|trw|http://asterisk.org) You can get it here : http://www.marccharbonneau.com/asterisk/mediaxphone.php Let me know what you think of it. hth ___ -- 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] 1.6.beta5 (format 0x40 (slin))
(alternative title - what did I do wrong? or suggestions to make this work) Thought I'd try 1.6 beta5 (and 1.4.18 didn't want to compile vpb /usr/lib/gcc/i386-redhat-linux/4.1.2/../../../../include/c++/4.1.2/i386-redhat-linux/bits/gthr-default.h:48: error: â does not name a type ) 1.6 did compile and almost works. 'cept it thinks the .gsm files are not played. from extensions.conf exten = s,2,BackGround(/var/lib/asterisk/sounds/en/vm-instructions.gsm) running asterisk -- Executing [EMAIL PROTECTED]:2] BackGround(vpb/0-3, /var/lib/asterisk/sounds/en/vm-instructions.gsm) in new stack [Mar 10 00:07:45] WARNING[23934]: file.c:557 ast_openstream_full: File /var/lib/asterisk/sounds/en/vm-instructions.gsm does not exist in any format [Mar 10 00:07:45] WARNING[23934]: file.c:856 ast_streamfile: Unable to open /var/lib/asterisk/sounds/en/vm-instructions.gsm (format 0x40 (slin)): No such file or directory [Mar 10 00:07:45] WARNING[23934]: pbx.c:7138 pbx_builtin_background: ast_streamfile failed on vpb/0-3 for /var/lib/asterisk/sounds/en/vm-instructions.gsm And the built in: -- Executing [EMAIL PROTECTED]:14] VoiceMail(vpb/0-0, 400) in new stack [Mar 10 00:07:36] WARNING[23934]: file.c:557 ast_openstream_full: File vm-intro does not exist in any format [Mar 10 00:07:36] WARNING[23934]: file.c:856 ast_streamfile: Unable to open vm-intro (format 0x40 (slin)): No such file or directory As you can see, the file exists. locate vm-intro /usr/share/asterisk/sounds/vm-intro.gsm /var/lib/asterisk/sounds/en/vm-intro.gsm ___ -- 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] SIP REFER Message, over NAT
Hi people, I have a few SPA-942 around, all of them work fine except one. The one behind NAT.. In every phone you can: * Pickup a Call on one of the line buttons, * Create a new call on another button * Press xferLx to join those to calls. This works everywhere except on the one behind NAT. After a lot of messing around with all the options possible I gave up and subscribed here... As of now, the only thing I've found out is that when you press the xferLx the SPA942 sends a REFER message to Asterisk, but there's one thing that seems wrong, the Via header. On every message that reaches Asterisk the Via header is: incoming: Via: SIP/2.0/UDP WAN IP:18363;branch=z9hG4bK-bb6024d6 outgoing: Via: SIP/2.0/UDP WAN IP:18363;branch=z9hG4bK-bb6024d6;received=WAN IP (either with rport or without it) On the REFER message the thing is different... incoming: Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK-37cbdb0 outgoing: Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK-37cbdb0;received=WAN IP Also, either with rport or without it. As you see, for some reason the REFER message has a wrong VIA header, I have no idea if the fault is on the SPA942 or Asterisk side. The problem is that with this Via header asterisk answers with 603 Declined (no dialog) to the REFER message and the SPA942 looses both legs of the call (well, it simply sends BYE for both). Any idea on how to fix the Via headers for the REFER message? I tried with nat=yes and nat=no and almost any possible combination of the SPA942 options so I don't paste the configs ;) Thanks, Marc -- http://www.marcfargas.com -- will be finished some day. signature.asc Description: Esta parte del mensaje está firmada digitalmente ___ -- 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 config file online editor
On Feb 19, 2008 10:44 AM, Anton Krall [EMAIL PROTECTED] wrote: Guys, Im looking for a good text file editor for asterisk config files that can be embedded on a web page for online editing (on an interface), any recommendations? You mean, something like this : http://www.voip-info.org/wiki/index.php?page=Asterisk+gui+phpconfig hth ___ -- 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] restart asterisk daily
I found that there will be a memory leak if asterisk running day by day without restart. Is it good to restart asterisk service daily? What is the better way to restart it daily like apache? Probably depends on the version of Asterisk, but I don't restart daily From one in production used daily with call recording, conference, etc : System uptime: 18 weeks, 5 days, 10 hours, 56 minutes, 37 seconds And the last reboot was because of a major power failure that lasted longer than the UPS could stand. The worst I've seen was a weekly restart of Asterisk, just to be on the safe side What version of Asterisk are you using ? ___ -- 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] [Softphones] ZoIPer vs. XLite?
Are other clients I should know about? http://www.zoiper.com/ http://www.counterpath.com/ Add to that list - Mozphone (http://mozphone.mozdev.org/) that can be installed in Firefox -Kiax : http://sourceforge.net/projects/kiax - shameless plugMy MediaX softphone : http://www.marccharbonneau.com/asterisk/mediaxphone.php/shameless plug - iaxcomm : http://iaxclient.sourceforge.net/iaxcomm/ - The one from Sokol associates : http://www.sokol-associates.com/?q=node/29 hth ___ -- 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] [Softphones] ZoIPer vs. XLite?
Marc, does your client play nicely with Vista? We've been having some problems with softphones that work fine in XP, but choke in Vista. I don't know, never tried it since I couldn't find a machine with enough power to run Vista decently ;) Try it and let me know how it goes. If it doesn't work, I will try to fix it. 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] Share accounts several AOR
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Good morning, Is it possible with asterisk to allow to share the same account on 2 different devices, for example I want both my fix phone and my wifi phone to ring in the same time. I want to do it without making ringroups... Any idea how to do it? Thanks -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHma77rxOjjFYWQtoRAv6+AKCXqImQPJK0NxXHZlJDu6BShelwJwCeKVtj AAPzlXluS9e3t1qPXqA6sPU= =fpsa -END PGP SIGNATURE- ___ -- 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] IAX softphone
Any one advise a good strong softphone that can work with IAX fine? samelessplugTry my softphone : http://www.marccharbonneau.com/asterisk/mediaxphone.php/samelessplug ___ -- 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] [IAX] Up-to-date list of soft- and hardphones?
Diax is probably the smallest Windows softphone. Add to that list Mozphone (http://mozphone.mozdev.org/) that can be installed in Firefox Kiax : http://sourceforge.net/projects/kiax shameless plugMy MediaX softphone : http://www.marccharbonneau.com/asterisk/mediaxphone.php/shameless plug iaxcomm : http://iaxclient.sourceforge.net/iaxcomm/ The one from Sokol associates : http://www.sokol-associates.com/?q=node/29 There is other ones also, Google is your friend As for a hardware IAX phone, I can't recommend one as I never tried one. hth ___ -- 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] macports testing of asterisk
Hi, I recently submitted to the macports project a portfile enabling MacOSX users to use the simple macports system to install asterisk. Until it is integrated into macports svn, I'm looking for users to try it out and find bugs/enhancements/suggestions. how to test the port by installing the portfile manually until it is integrated into the macports svn: mkdir -p ~/ports/net/asterisk cd ~/ports/net/asterisk wget 'http://trac.macosforge.org/projects/macports/attachment/ticket/13749/Portfile?format=raw' -O Portfile cd ~/ports sudo vi /opt/local/etc/macports/sources.conf add the following line: file:///Users/YOURUSERID/ports portindex sudo port install asterisk Please report success or issues directly to me. Thanks, Marc. ___ --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] MWI error
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Good Morning, My problem was that the context wasn't the same in my voicemail.conf and in my sip.conf!! One was 'default' and the other 'device' I have put 'default' everywhere and it's working! Have a nice day Jared Smith a écrit : On Tue, 2007-12-04 at 17:20 +0100, Marc LEURENT wrote: It's just that I received SIP notify message saying that there is nothing in the voicemail even when there is a message... Do you have a mailbox defined for the SIP device in sip.conf? If you don't, Asterisk has no way of matching up a mailbox to a particular SIP device. -Jared Smith ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHVmcvN4+o+2LtdFwRAkdSAJ9KPkr9NGc9nm+wIFGUofcE4nxQnACfRJeL HakgTsDpHM7QCCyvzPI0440= =J5cK -END PGP SIGNATURE- ___ --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] MWI error
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 It's just that I received SIP notify message saying that there is nothing in the voicemail even when there is a message... my voicemail.conf [default] ; Define maximum number of messages per folder for a particular context. ;maxmsg=50 ; Mailboxes may be organized into multiple contexts for ; voicemail virtualhosting ; 6710 = 1234,Compte Test 0,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=yes|saycid=yes|envelope=yes|delete=no Alex Balashov a écrit : Sorry, not sure I understand the question. What is the problem here? On Mon, 3 Dec 2007, Marc LEURENT wrote: Good evening, I have something strange, I have unread message in my voicemail box but the SIP NOTIFY that are received by my telephone are like: whereas there is voice messages inside! Any idea how to solve that? Thanks PS: I'm using asterisk 1.4.13 + Freepbx # U 192.168.95.235:5060 - 192.168.95.73:5060 NOTIFY sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0. v: SIP/2.0/UDP 192.168.95.235:5060;branch=z9hG4bK114bbd0e;rport. f: Unknown sip:[EMAIL PROTECTED];tag=as5087d720. t: sip:[EMAIL PROTECTED]:5060;user=phone. m: sip:[EMAIL PROTECTED]. i: [EMAIL PROTECTED] CSeq: 102 NOTIFY. User-Agent: Asterisk PBX. Max-Forwards: 70. o: message-summary. c: application/simple-message-summary. l: 94. . Messages-Waiting: no. Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 0/0 (0/0). ___ - --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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHVX5jN4+o+2LtdFwRAicZAKCwjAojZxq6gbF2+qvyUozYteBwMACfZq51 WqddUJCEAI7Q18V3ROv0FVk= =tKYm -END PGP SIGNATURE- ___ --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] MWI error
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Good evening, I have something strange, I have unread message in my voicemail box but the SIP NOTIFY that are received by my telephone are like: whereas there is voice messages inside! Any idea how to solve that? Thanks PS: I'm using asterisk 1.4.13 + Freepbx # U 192.168.95.235:5060 - 192.168.95.73:5060 NOTIFY sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0. v: SIP/2.0/UDP 192.168.95.235:5060;branch=z9hG4bK114bbd0e;rport. f: Unknown sip:[EMAIL PROTECTED];tag=as5087d720. t: sip:[EMAIL PROTECTED]:5060;user=phone. m: sip:[EMAIL PROTECTED]. i: [EMAIL PROTECTED] CSeq: 102 NOTIFY. User-Agent: Asterisk PBX. Max-Forwards: 70. o: message-summary. c: application/simple-message-summary. l: 94. . Messages-Waiting: no. Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 0/0 (0/0). -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHVEeaN4+o+2LtdFwRAhfGAJ4/iL4yG0xm5XBaYLUxGzpgKitGNwCfREV+ H9wJ6bD+ITOBDoKm2gstEQQ= =3MmR -END PGP SIGNATURE- ___ --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] G729 on wrong bus
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Just download the g729 module that fits your hardware at http://downloads.digium.com/pub/telephony/codec_g729/ and follow the README: http://downloads.digium.com/pub/telephony/codec_g729/README PS: do a 'cat /proc/cpuinfo' to know what it your processor Good luck broadband Voice a écrit : Hi, Can anyone assist me in resolving this problem? I installed the G729 on a 32 and just found out that the server is 64. 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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHTS3nN4+o+2LtdFwRApC+AJ9uIY4OOVVYvoUr6f3AIAcpMrIZPgCdFl4/ jb20I7KhhfPssGWFgSXK+5w= =7res -END PGP SIGNATURE- ___ --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] 1 FXS module / PCI express
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Good morning, I would like to find a simple PCI express card with only one FXS module, do you know where I can find such a card? Thanks -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHTTDzN4+o+2LtdFwRAvh0AJ434R5EEYLAfywDsSiCylw6nCMqVQCgmjSP IV4/QrSNiLsqcMa0k6yv/j8= =Wqy0 -END PGP SIGNATURE- ___ --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] Best Prepaid Application?
Thank you for your answer! I'm going to try it! Have a nice day Mindaugas Kezys a écrit : You can try MOR FREE - it has nice gui and is very fast. LiveCD is available: http://www.kolmisoft.com/mor/content/view/83/95/ It is covered in extensive manual: http://www.kolmisoft.com/mor/component/option,com_remository/Itemid,40/func, fileinfo/id,25/ And yes - it's FREE as name suggests. Regards/Pagarbiai, Mindaugas Kezys Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marc LEURENT Sent: Friday, November 23, 2007 7:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Best Prepaid Application? Good evening, Have you got any idea which prepaid application will be the best to do simple prepaid calls with a MySQL storage...? PS: I have a compiled by hand Asterisk 1.4.13 on a Debian Etch 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 ___ --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] Asterisk B2BUA patch useful??
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, Is the asterisk B2BUA patches useful anymore?? I'm trying to set a prepaid SIP network and the only way seems to get through a patched asterisk with B2BUA functions.. The patches failed, Hunk + problems: I have repaired them, but is it very useful?? Thanks -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHSsB+N4+o+2LtdFwRAhcsAJ9SBj/AMVka+tcs068BuSIksyuVsQCgy35r 8lcQv58TD45eAAxCKrAU75M= =NZ4+ -END PGP SIGNATURE- ___ --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] Best Prepaid Application?
Good evening, Have you got any idea which prepaid application will be the best to do simple prepaid calls with a MySQL storage...? PS: I have a compiled by hand Asterisk 1.4.13 on a Debian Etch 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] route INVITE sip:[EMAIL PROTECTED]
We are using 2 different incoming trunks. The first one is alsion.com and is sending INVITE with phone number in the INVITE line whereas plugandtel put the callee number only inside the To: Section. Marco Mouta a écrit : Could you describe in detail how did you fall into this situation, I mean the real example which SIP phone sends this invite? Is registered in asterisk? it is a non-registered sip phone trying to dial a sip user at your * box? If this is an issue with a specific hardware outside of your asterisk, may be something not well configured ... describe it a bit more in detail. If you don't have anyworkaround for this Invite format I would use OpenSER in front of Asterisk to handle this invites and replace to SIP URI with info from the tag TO: ... Any way if you provide more details may be someone in the Mailing list is able to help u out;) Best regards MoutaPT On Nov 13, 2007 6:14 PM, Marc LEURENT [EMAIL PROTECTED] wrote: Good evening! I was wondering one thing, I'm using freepbx to configure my asterisk server and I have a problem with some inbound calls. When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an inbound route! It matches a DID number. How can I route an INVITE sip:[EMAIL PROTECTED] The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it (there is only the number to be reached in the To: section) # U 217.36.112.145:5060 - 192.168.95.235:5060 INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0. Allow: UPDATE,REFER,INFO. Call-ID: [EMAIL PROTECTED] Contact: sip:217.66.118.145:5060. Content-Type: application/sdp. CSeq: 34878212 INVITE. From: 0614740696 sip:[EMAIL PROTECTED];user=phone;tag=02975-US-0223ae6e-67d6c4495. Max-Forwards: 31. To: sip:[EMAIL PROTECTED];user=phone. User-Agent: Cirpack/v4.41c (gw_sip). Via: SIP/2.0/UDP 217.36.112.145:5060;branch=z9hG4bK-744D-33B812. Content-Length: 303. . Whereas with this one I can do it! (there is a number in the INVITE) # U 87.98.202.114:5060 - 192.168.95.235:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP 87.98.202.114:5060;branch=z9hG4bK1fd2c6b4;rport. From: 0158136741 sip:[EMAIL PROTECTED];tag=as25391ca7. To: sip:[EMAIL PROTECTED]. Contact: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Tue, 13 Nov 2007 18:07:00 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 233. . ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] route INVITE sip:[EMAIL PROTECTED]
Good evening! I was wondering one thing, I'm using freepbx to configure my asterisk server and I have a problem with some inbound calls. When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an inbound route! It matches a DID number. How can I route an INVITE sip:[EMAIL PROTECTED] The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it (there is only the number to be reached in the To: section) # U 217.36.112.145:5060 - 192.168.95.235:5060 INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0. Allow: UPDATE,REFER,INFO. Call-ID: [EMAIL PROTECTED] Contact: sip:217.66.118.145:5060. Content-Type: application/sdp. CSeq: 34878212 INVITE. From: 0614740696 sip:[EMAIL PROTECTED];user=phone;tag=02975-US-0223ae6e-67d6c4495. Max-Forwards: 31. To: sip:[EMAIL PROTECTED];user=phone. User-Agent: Cirpack/v4.41c (gw_sip). Via: SIP/2.0/UDP 217.36.112.145:5060;branch=z9hG4bK-744D-33B812. Content-Length: 303. . Whereas with this one I can do it! (there is a number in the INVITE) # U 87.98.202.114:5060 - 192.168.95.235:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP 87.98.202.114:5060;branch=z9hG4bK1fd2c6b4;rport. From: 0158136741 sip:[EMAIL PROTECTED];tag=as25391ca7. To: sip:[EMAIL PROTECTED]. Contact: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Tue, 13 Nov 2007 18:07:00 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 233. . ___ --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] G729a codecs + Asterisk 1.4.11
Of course, use the codec for the pentium 4!! bilal ghayyad a écrit : Dear Marc; Thanks a lot for your kindly help. My output of the command cat /proc/cpuinfo is: [EMAIL PROTECTED] /]# cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Pentium(R) 4 CPU 3.40GHz stepping: 1 cpu MHz : 3391.901 cache size : 1024 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr bogomips: 6787.39 clflush size: 64 So what means I have to download codec_g729a_v32_pentium4m.tar.gz ? But the output of the command (from Asterisk CLI): CLI core show version is: Asterisk SVN-branch-1.4-r72556 built by root @ localhost.localdomain on a i686 running Linux on 2007-06-30 13:08:08 UTC I tried this codec: codec_g729a_v32_i686.tar.gz but when I type show modules like 72 then it gives me the following (that doesnot contain g729.a): localhost*CLI show modules like 72 Module Description Use Count format_g729.so Raw G729 data 0 format_g723.so G.723.1 Simple Timestamp File Format 0 codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 4 modules loaded So can u advise? Regards Bilal --- Marc LEURENT [EMAIL PROTECTED] wrote: To know your architecture, use the cmd: cat /proc/cpuinfo After try to start to use the version below (i686): http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-32/codec_g729a_v32_i686.tar.gz Good luck bilal ghayyad a écrit : Dear Marc; I readed your email about the codec G729a and I am now also need to install the codec on my Asterisk. I typed from Asterisk CLI: core show version and I got the following: Asterisk SVN-branch-1.4-r72556 built by root @ localhost.localdomain on a i686 running Linux on 2007-06-30 13:08:08 UTC So I beleive that my processor is i686, correct? But I am not able to know which one to download: The x86-32 or x86-64 ? Can you please advise. Also, the nocona or the opteron versions? Regards Bilal --- Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64/codec_g729a_v32_nocona.tar.gz codec I have registered my license, copied the codec_g729a.so into the /usr/lib/asterisk/modules folder and restarted my asterisk But on the CLI when I type asterisk*CLI show modules like 72 Module Description Use Count codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g729.so Raw G729 data 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_g723.so G.723.1 Simple Timestamp File Format 0 The codec_g729a.so doesn't appear.. Any idea how to solve the problem. Thanks Best Regards, Marc LEURENT __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729a codecs + Asterisk 1.4.11
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 To know your architecture, use the cmd: cat /proc/cpuinfo After try to start to use the version below (i686): http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-32/codec_g729a_v32_i686.tar.gz Good luck bilal ghayyad a écrit : Dear Marc; I readed your email about the codec G729a and I am now also need to install the codec on my Asterisk. I typed from Asterisk CLI: core show version and I got the following: Asterisk SVN-branch-1.4-r72556 built by root @ localhost.localdomain on a i686 running Linux on 2007-06-30 13:08:08 UTC So I beleive that my processor is i686, correct? But I am not able to know which one to download: The x86-32 or x86-64 ? Can you please advise. Also, the nocona or the opteron versions? Regards Bilal --- Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64/codec_g729a_v32_nocona.tar.gz codec I have registered my license, copied the codec_g729a.so into the /usr/lib/asterisk/modules folder and restarted my asterisk But on the CLI when I type asterisk*CLI show modules like 72 Module Description Use Count codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g729.so Raw G729 data 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_g723.so G.723.1 Simple Timestamp File Format 0 The codec_g729a.so doesn't appear.. Any idea how to solve the problem. Thanks Best Regards, Marc LEURENT __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHHaLSqjpLE0HiOBYRAizdAJ9r8Hm83u/EMDBeaFCseW/XofIIYwCfbKpk xWjhS4+xRj5G9HpQYAEfwhY= =0rDl -END PGP SIGNATURE- ___ --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] Alert_INFO x2 = 400 Bad Request
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Good evening,
I have something strange, when I add an ALERT_INFO variable to a ring group,
the invite generated contains 2 lines with Alert-Info and my phones return a
400 Bad Request...
I've checked in my config files, there is only one line with
Set(__ALERT_INFO.
Any idea??
PS: I'm using Asterisk Asterisk 1.4.13-BRIstuffed-0.4.0-test4
Thanks
Alert-Info: Ringer-2.
Alert-Info: Ringer-2.
exten = 6800,1,Macro(user-callerid,)
exten = 6800,n,GotoIf($[foo${BLKVM_OVERRIDE} = foo]?skipdb)
exten = 6800,n,GotoIf($[${DB(${BLKVM_OVERRIDE})} = TRUE]?skipov)
exten = 6800,n(skipdb),Set(__NODEST=)
exten = 6800,n,Set(__BLKVM_OVERRIDE=BLKVM/${EXTEN}/${CHANNEL})
exten = 6800,n,Set(__BLKVM_BASE=${EXTEN})
exten = 6800,n,Set(DB(${BLKVM_OVERRIDE})=TRUE)
exten = 6800,n(skipov),Set(RRNODEST=${NODEST})
exten = 6800,n(skipvmblk),Set(__NODEST=${EXTEN})
exten = 6800,n,Set(__ALERT_INFO=Ringer-1)
exten = 6800,n,Set(RecordMethod=Group)
exten = 6800,n,Macro(record-enable,6740,${RecordMethod})
exten = 6800,n,Set(RingGroupMethod=ringall)
exten = 6800,n(DIALGRP),Macro(dial,7,${DIAL_OPTIONS},6740)
exten = 6800,n,Set(RingGroupMethod=)
exten = 6800,n,GotoIf($[foo${RRNODEST} != foo]?nodest)
exten = 6800,n,Set(__NODEST=)
exten = 6800,n,dbDel(${BLKVM_OVERRIDE})
exten = 6800,n,Goto(ext-group,6799,1)
exten = 6800,n(nodest),Noop(SKIPPING DEST, CALL CAME FROM Q/RG: ${RRNODEST})
#
U 192.168.95.235:5060 - 192.168.95.73:5060
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 192.168.95.235:5060;branch=z9hG4bK7f1bf341;rport.
From: 0614730696 sip:[EMAIL PROTECTED];tag=as6aaa622f.
To: sip:[EMAIL PROTECTED]:5060;user=phone.
Contact: sip:[EMAIL PROTECTED].
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Date: Thu, 11 Oct 2007 17:00:58 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Alert-Info: Ringer-2.
Alert-Info: Ringer-2.
Content-Type: application/sdp.
Content-Length: 266.
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[asterisk-users] G729a codecs + Asterisk 1.4.11
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64/codec_g729a_v32_nocona.tar.gz codec I have registered my license, copied the codec_g729a.so into the /usr/lib/asterisk/modules folder and restarted my asterisk But on the CLI when I type asterisk*CLI show modules like 72 Module Description Use Count codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g729.so Raw G729 data0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_g723.so G.723.1 Simple Timestamp File Format 0 The codec_g729a.so doesn't appear.. Any idea how to solve the problem. Thanks Best Regards, Marc LEURENT -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHDNxdqjpLE0HiOBYRAug5AJ4qjE57UcgHEsmAVQFwPSyMn/dyogCeP3qG UKXWhR9ebm2iw2Ao8VLuSEk= =7O/k -END PGP SIGNATURE- ___ --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] G729a codecs + Asterisk 1.4.11
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ok, I've downloaded the i386 module and it works, I have the module loaded... Thanks for the command!! Rafael Canchola a écrit : Hi: You can check the next command: show g729 and you should see some like this 0/0 encoders/decoders of 2 licensed channels are currently in use or the command show translation or check the asterisk log may be the module is not for you processor version. Best Regards At 09:06 a.m. 10/10/2007, you wrote: Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64/codec_g729a_v32_nocona.tar.gz codec I have registered my license, copied the codec_g729a.so into the /usr/lib/asterisk/modules folder and restarted my asterisk But on the CLI when I type asterisk*CLI show modules like 72 Module Description Use Count codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g729.so Raw G729 data0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_g723.so G.723.1 Simple Timestamp File Format 0 The codec_g729a.so doesn't appear.. Any idea how to solve the problem. Thanks Best Regards, Marc LEURENT ___ - --Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users */ Rafael/*/Canchola //*Product Development Engineer*/*, Fonet*Global Inc. [EMAIL PROTECTED] http://www.fonetglobal.com http://www.fonetglobal.com/*Ph. *+ 52 800 022 10 21 ext. 214 + 52 442 167 08 00 *VoIP* 523663899 *d00d! *cyberalph ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHDO2XqjpLE0HiOBYRAtcTAJ9YJ8qC83ZxC0+kvf3hfAWvb0/FmgCfb2te F8vtQ07kypElJEsokR1XrD8= =lUtS -END PGP SIGNATURE- ___ --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] How to order audio codecs...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have license for g729a audio codecs and I would like user to use them and when the limit of 10 is reached, I would like the others to use ulaw... Do youu know how to do it... I have put: allow=g729,ulaw disallow=all But ulaw is always chosen Have a nice day -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHDPePqjpLE0HiOBYRAvSWAJ9Z7gJMDuTw9EcL5of35SmF1slwIwCeM8n/ MfjqNU/3gkdLwKqo1tN5yV8= =3oU/ -END PGP SIGNATURE- ___ --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] Backuping VoIP provider with PRI
Thx Steve! Steve Totaro wrote: Qualify=yes? Thanks, Steve Marc Patino Gómez wrote: Hi Adam, thanks for your quick answer, I try your tip but the problem persist, so... It seems not to be a dns problem Asterisk executes the Dial command and it tries to reach the VoIP provider until timeout, in * console appears: Called [EMAIL PROTECTED] Anybody knows howto make dial command don't wait until timeout when the provider host is unrechable? Cheers, Marc Adam KOSA wrote: Marc Patino Gómez wrote: in most cases it works well but, if my internet connection is down Asterisk tries to Dial voipprovider, but it can't resolve the dns name, so it waits 60 seconds to jump to the following priority... Any ideas to solve this problem? I can't use the IP of the provider (it has a pool of servers), I try to use dnsmgr without solving the isue Why don't you fill the ip addresses to your /etc/hosts file? In that way lookups won't need any dns resolving and still could keep the load balancing by having multiple ip addresses to the same SIP hostname. regards Adam ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Backuping VoIP provider with PRI
Hi list,
My Asterisk config for outgoing calls is the following:
exten = s,1,Dial(SIP/[EMAIL PROTECTED],60,g)
exten = s,n,GotoIf($[\${ANSWEREDTIME}\ = \\]?pri:hang)
exten = s,n(pri),NoOp(Problems with voip provider trying PRI)
exten = s,n,Dial(Zap/g2/${MACRO_EXTEN},60,g)
exten = s,n(hang),HangUp
in most cases it works well but, if my internet connection is down
Asterisk tries to Dial voipprovider, but it can't resolve the dns name,
so it waits 60 seconds to jump to the following priority...
Any ideas to solve this problem? I can't use the IP of the provider (it
has a pool of servers), I try to use dnsmgr without solving the isue
Thanks in advance,
Marc
PD: I have used more sophisticate configs using DIALSTATUS variable, but
the problem persists
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Re: [asterisk-users] Backuping VoIP provider with PRI
Hi Adam, thanks for your quick answer, I try your tip but the problem persist, so... It seems not to be a dns problem Asterisk executes the Dial command and it tries to reach the VoIP provider until timeout, in * console appears: Called [EMAIL PROTECTED] Anybody knows howto make dial command don't wait until timeout when the provider host is unrechable? Cheers, Marc Adam KOSA wrote: Marc Patino Gómez wrote: in most cases it works well but, if my internet connection is down Asterisk tries to Dial voipprovider, but it can't resolve the dns name, so it waits 60 seconds to jump to the following priority... Any ideas to solve this problem? I can't use the IP of the provider (it has a pool of servers), I try to use dnsmgr without solving the isue Why don't you fill the ip addresses to your /etc/hosts file? In that way lookups won't need any dns resolving and still could keep the load balancing by having multiple ip addresses to the same SIP hostname. regards Adam ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Dell SC1430 + Digium TE110P = Digital Noise in PRI SOLVED!!
Hi list, After talking with Digium, they shipped to me a TE120P, this card with his modern chipset has solved the noise issue. I'm very happy with Digium support, specially with Rod and Russell. This issue shows me that behind Digium Inc there are people who helps their customers with a great support. Thanks to all, Marc Matthew Fredrickson wrote: Arthur Miller wrote: The Digium cards are known to steal IRQ's. The Sangoma cards do not Not to appear defensive, but that is a technically inaccurate and also technically ambiguous statement. To correct it, there used to be a potential problem related to using the TE2xxP/TE4xxP cards relating to IRQ sharing which was fixed by a driver update. That is now resolved, and there shouldn't be any further issues. A considerable portion of the IRQ problems are an urban legend, a sort of scapegoat to point at. However, I would like to say that if anyone *does* have any problems relating to this, Digium and I personally are *very* interested in correcting them. We want to make sure that you trust our products, and want to stand behind our ability to support that. We have had some growing pains along the way, but we are *very* interested in making sure our hardware works to your and our other customers' satisfaction, and certainly stands up for itself in the face of competition. The Asterisk community is very important to us, and your perception of our products is crucial to our ability to afford to better support you and also forward the development of Asterisk. If you do have a problem, please contact technical support so that it can be fixed as soon as possible. ___ --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] Dell SC1430 + Digium TE110P = Digital Noise in PRI
Hi Russell, First of all, let me tell that in my company only buy Digium Cards, because: - Is the company founded by Mark Spencer, and buying Digium hardware is a way to support Asterisk (in my opinion) - Since today I only can tell good things about Digium: good support to the comunity, good care to their customers... somebody can tell me that is romantic ... but this is my opinion. Now I'm very happy to tell to the list the following, Digium contacted to me to test the TE120P, as you told, its modern PCI interface will solve my issue ( I hope :) ). After my experience with this new card I will post my feelings and results to the list. Thanks to all, Marc PD: I hope after solve my issue, I will wear my Asterisk t-shirt (with Digium logo on its back) as same as proud that I wear my Debian t-shirt ;) Russell Bryant wrote: Marc Patino Gómez wrote: I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with lspci -vb command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same Digium card and there is no noise in PRI. Any advice to solve the problem with Dell box? Did you ever contact Digium technical support to give them a chance to fix your problem? It is really disappointing to see people go with another vendor without even giving us a chance to resolve your issue. If the TE110P will not work out for you, Digium will trade it for a TE120P. The 120 is the replacement for the 110 which uses a far superior PCI interface developed at Digium instead of the TigerJet, which has been the cause of compatability issues in the past. Very soon, the TigerJet part will no longer be in use in any of the Digium cards. ___ --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] where is 1.4.12?
Yes.. OUR rollouts work fine, because we use a version of asterisk that we are comfortable with. However, I'm talking about when we do consulting for someone who has installed their own asterisk and then they have some issues with it... This is the problem to use the last release of software A good sysadmin never will put the latest releases in a critical production environment, free software and also proprietary software. Asterisk is free software and in my opinion is a task of all (Community and Digium) to make it better. Please, don't discuss, fix bugs ;) ___ --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] Dell SC1430 + Digium TE110P = Digital Noise in PRI
Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with lspci -vb command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same Digium card and there is no noise in PRI. Any advice to solve the problem with Dell box? Regards, Marc ___ --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] Dell SC1430 + Digium TE110P = Digital Noise in PRI
Hi Steve, All my cards are Digium, I tried diferent Digium cards and I had the same problem. Regards, Marc Steve Totaro wrote: Marc Patino Gómez wrote: Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with lspci -vb command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same Digium card and there is no noise in PRI. Any advice to solve the problem with Dell box? Regards, Marc Try a Sangoma card? Thanks, Steve Totaro ___ --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] Dell SC1430 + Digium TE110P = Digital Noise in PRI
Hi Steve, Thanks for your advice, I will order a Sangoma card and test the box. A part from this, you know any other point to recomend Sangoma cards versus Digium cards? Many thanks, Marc Steve Totaro wrote: That is why I suggested Sangoma. Ask them if you can return it if it does not fix your problem. It is alot easier than disabling things in BIOS and hunting for the elusive noises. Digium would have you believe that the problem is the Dell box but if a Sangoma card works perfectly in the same box, then where is the actual problem? Anyways, if you are set on Digium, call their support and give them SSH. They may be your best bet in fixing the issue. Thanks, Steve Marc Patino Gómez wrote: Hi Steve, All my cards are Digium, I tried diferent Digium cards and I had the same problem. Regards, Marc Steve Totaro wrote: Marc Patino Gómez wrote: Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with lspci -vb command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same Digium card and there is no noise in PRI. Any advice to solve the problem with Dell box? Regards, Marc Try a Sangoma card? Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Dell SC1430 + Digium TE110P = Digital Noise in PRI
Hi John, thanks for this usefull info Marc John Novack wrote: Marc Patino Gómez wrote: Hi Steve, Thanks for your advice, I will order a Sangoma card and test the box. A part from this, you know any other point to recomend Sangoma cards versus Digium cards? Many thanks, Marc 5 year warranty, to name one. Sangoma says their cards will work in ALL modern machines. If they can't make it work ( never seen that ) they will refund. If you have problems, and you give them SSH, they will fix it. John Novack Steve Totaro wrote: That is why I suggested Sangoma. Ask them if you can return it if it does not fix your problem. It is alot easier than disabling things in BIOS and hunting for the elusive noises. Digium would have you believe that the problem is the Dell box but if a Sangoma card works perfectly in the same box, then where is the actual problem? Anyways, if you are set on Digium, call their support and give them SSH. They may be your best bet in fixing the issue. Thanks, Steve Marc Patino Gómez wrote: Hi Steve, All my cards are Digium, I tried diferent Digium cards and I had the same problem. Regards, Marc Steve Totaro wrote: Marc Patino Gómez wrote: Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with lspci -vb command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same Digium card and there is no noise in PRI. Any advice to solve the problem with Dell box? Regards, Marc Try a Sangoma card? Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Post voicemail processing.
This 2 line code is doing what I wanted. exten = 200,1,voicemail(200) exten = 200,2,Hangup What I've been told is that they want the 20 year old phone system to light up the message bulb. (yea, a filament bulb, not an LED) To do this you pick up on the line that goes into Asterisk and do a: exten = 200,1,SendDTMF(200w#86) But I don't know the path to take to get that to happen after the hangup() - which is needed to close down the voicemail() call. I've tried: exten = 200,1,voicemail(200) exten = 200,2,Hangup exten = 200,3,Wait(2) exten = 200,4,Answer exten = 200,5,SendDTMF(200w#86) exten = 200,6,Hangup (go to voicemail, hangup, wait 2 seconds, lift hook, make noise, hangup) and: exten = 200,1,voicemail(200) exten = 200,2,softHangup exten = 200,3,Wait(2) exten = 200,4,Answer exten = 200,5,SendDTMF(200w#86) exten = 200,6,Hangup (go to voicemail, hangup hook but keep processing, wait 2 seconds, lift hook, make noise, hangup to end process) ___ --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] trying to get vpb to compile
So I've got a Voicetronix card and it looks like the kernel driver works. Other than the 0's for ID info. vpb: Driver Version = 4.0 vpb: major = 251 vpb: tmp [0xfc8fec00] dev-res3 [0xfc8fec00] vpb: tmp [0xfc8c] dev-res2 [0xfc8c] vpb: 1WS Write cycle vpb: Manufactured 00/00/ vpb: Card version 00.00 vpb: Serial number vpb: Setting up udev... vpb:1 V4PCI's detected on PCI bus [EMAIL PROTECTED] asterisk-1.4.4]# make menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. [CXX] chan_vpb.cc - chan_vpb.oo chan_vpb.cc:382: error: invalid conversion from \xe2 to \xe2 chan_vpb.cc:410: error: invalid conversion from \xe2 to \xe2 chan_vpb.cc: In function \xe2: chan_vpb.cc:1530: warning: dereferencing type-punned pointer will break strict-aliasing rules chan_vpb.cc: In function \xe2: chan_vpb.cc:2637: error: \xe2 has no member named \xe2 chan_vpb.cc:2671: warning: comparison between signed and unsigned integer expressions make[1]: *** [chan_vpb.oo] Error 1 make: *** [channels] Error 2 [EMAIL PROTECTED] asterisk-1.4.4]# cd ../as*6 [EMAIL PROTECTED] asterisk-1.4.6]# make menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. [CXX] chan_vpb.cc - chan_vpb.oo chan_vpb.cc:382: error: invalid conversion from \xe2 to \xe2 chan_vpb.cc:410: error: invalid conversion from \xe2 to \xe2 chan_vpb.cc: In function \xe2: chan_vpb.cc:1530: warning: dereferencing type-punned pointer will break strict-aliasing rules chan_vpb.cc: In function \xe2: chan_vpb.cc:2637: error: \xe2 has no member named \xe2 make[1]: *** [chan_vpb.oo] Error 1 make: *** [channels] Error 2 [EMAIL PROTECTED] asterisk-1.4.6]# Ok, so I'm seeing less errors going from 1.4.4 to 1.4.6 so lets try the trunk via SVN [CXX] chan_vpb.cc - chan_vpb.oo chan_vpb.cc:3055:9: error: macro AST_MODULE_INFO passed 7 arguments, but takes just 6 chan_vpb.cc:379: error: invalid conversion from \xe2 to \xe2 chan_vpb.cc:379: sorry, unimplemented: non-trivial designated initializers not supported chan_vpb.cc:379: sorry, unimplemented: non-trivial designated initializers not supported chan_vpb.cc:379: sorry, unimplemented: non-trivial designated initializers not supported chan_vpb.cc:379: sorry, unimplemented: non-trivial designated initializers not supported chan_vpb.cc:379: sorry, unimplemented: non-trivial designated initializers not supported chan_vpb.cc:379: sorry, unimplemented: non-trivial designated initializers not supported chan_vpb.cc:379: sorry, unimplemented: non-trivial designated initializers not supported chan_vpb.cc:379: error: uninitialized const member \xe2 chan_vpb.cc:379: error: uninitialized const member \xe2 chan_vpb.cc:407: error: invalid conversion from \xe2 to \xe2 chan_vpb.cc:407: sorry, unimplemented: non-trivial designated initializers not supported chan_vpb.cc:407: sorry, unimplemented: non-trivial designated initializers not supported chan_vpb.cc:407: sorry, unimplemented: non-trivial designated initializers not supported chan_vpb.cc:407: sorry, unimplemented: non-trivial designated initializers not supported chan_vpb.cc:407: sorry, unimplemented: non-trivial designated initializers not supported chan_vpb.cc:407: sorry, unimplemented: non-trivial designated initializers not supported chan_vpb.cc:407: sorry, unimplemented: non-trivial designated initializers not supported chan_vpb.cc:407: error: uninitialized const member \xe2 chan_vpb.cc:407: error: uninitialized const member \xe2 chan_vpb.cc: In function \xe2: chan_vpb.cc:1502: error: invalid conversion from \xe2 to \xe2 chan_vpb.cc:1526: warning: dereferencing type-punned pointer will break strict-aliasing rules chan_vpb.cc: In function \xe2: chan_vpb.cc:1546: error: invalid conversion from \xe2 to \xe2 chan_vpb.cc: In function \xe2: chan_vpb.cc:2628: error: \xe2 has no member named \xe2 chan_vpb.cc: In function \xe2: chan_vpb.cc:3032: error: \xe2 was not declared in this scope chan_vpb.cc: At global scope: chan_vpb.cc:3051: error: expected constructor, destructor, or type conversion before \xe2 token make[1]: *** [chan_vpb.oo] Error 1 make: *** [channels] Error 2 ___ --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 + Legacy PBX
Hi all,
I have a isue with a Siemens Hicom conected to my asterisk, here is the
scheme:
Telco Asterisk --- Legacy PBX --- Legacy phones
The asterisk box has a TE210 (one PRI conected to Telco another PRI
conected to Siemens)
Everything works ok, but when I make an international call from legacy
phones to the telco, for example: 0034934452740, the Siemens only sends
to Asterisk the three first numbers 003.
Here is my config in extensions.conf:
[incoming-siemens]
exten = _X.,1,NoOP
exten = _X.,n,Dial(Zap/g2/${EXTEN})
exten = _X.,n,Hangup
[incoming-telco]
exten = _X.,1,NoOP
exten = _X.,n,Dial(Zap/g1/${EXTEN})
exten = _X.,n,Hangup
The other calls works great, incoming calls and outgoing calls. Any help
will be very apreciated, I'm a newbie doing this kind of asterisk
config, so any advice will be helpful.
Best regards,
Marc
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[asterisk-users] None random SIP channel names
Hello, In Asterisk 1.4.4, the SIP channel names (SIP//peer/-/id/) do not seem to be unique. That means that the id associated to a peer is not random. Is that normal ? Because other asterisk versions give random id for each generated SIP channels of a peer. Regards, - marc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need a RTP/SIP Proxy to be used as SBC (Session Border Controller)
Thanks, I already found these names, but maybe I missed some ! Thanks again, JM On 5/14/07, Yossi Ben Hagai [EMAIL PROTECTED] wrote: Check rtpproxy from portone for media proxy and nat traversal. http://www.voip-info.org/wiki/view/Portaone+rtpproxy another option is the MediaProxy from AG projects: http://www.voip-info.org/wiki-MediaProxy Joss. On 5/11/07, Jean-Marc Salsa [EMAIL PROTECTED] wrote: Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need a RTP/SIP Proxy to be used as SBC (Session Border Controller)
Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIPGetHeader in Asterisk 1.4
Hi to all, I recently tried to upgrade my Asterisk 1.2 to 1.4. I use quite extensively SIPGetHeader cmd in my Dialplan. But this application is not found in 1.4.2, and I do not see it in 1.4.4code either ??? I could find indeed SIPAddHeader in code. BUT Where did SIPGetHeader go ? any new cmd replacing this one ? Thanks, Jean-Marc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPGetHeader in Asterisk 1.4
Thanks ! On 5/2/07, Manu Mehta [EMAIL PROTECTED] wrote: Hi, You can use function SIP_HEADER instead. See http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header *Manu Mehta* * * *A R I C E N T* Plot-17, Sector 18, Gurgaon 122015, Haryana, India Main +91.124.4095888 x3274 Fax +91.124.4095912 *Jean-Marc Salsa [EMAIL PROTECTED]* Sent by: [EMAIL PROTECTED] 05/02/2007 07:03 PM Please respond to Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com cc Subject [asterisk-users] SIPGetHeader in Asterisk 1.4 Hi to all, I recently tried to upgrade my Asterisk 1.2 to 1.4. I use quite extensively SIPGetHeader cmd in my Dialplan. But this application is not found in 1.4.2, and I do not see it in 1.4.4code either ??? I could find indeed SIPAddHeader in code. BUT Where did SIPGetHeader go ? any new cmd replacing this one ? Thanks, Jean-Marc___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *** Aricent-Unclassified *** DISCLAIMER: This message is proprietary to Aricent and is intended solely for the use of the individual to whom it is addressed. It may contain privileged or confidential information and should not be circulated or used for any purpose other than for what it is intended. If you have received this message in error, please notify the originator immediately. If you are not the intended recipient, you are notified that you are strictly prohibited from using, copying, altering, or disclosing the contents of this message. Aricent accepts no responsibility for loss or damage arising from the use of the information transmitted by this email including damage from virus. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Returning different SIP Hangup Cause
Hi, I would like to return different values/cause to another SIP Server with Hangup cmd. I tried to put different values in Hangup(xx) ... but it always returns the same value ! How can I send back different error cause ? Thanks, Jean-Marc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming SIP call
Well thanks for answering, When I test, I use my GSM and call the number my provider gives me. How often it works or not, I didn't make test like 10 calls per hour for a pretty long time so I can't exactly tell. When I test, well sometimes it works great, sometime, the incoming call is redirected to an phone that is connected on my DSL box. I didn't see the error message SIP/2.0 403 not registered, but in that case: 1) I can make a call from asterisk to a gsm call (so It goes IAX phone = asterisk = SIP provider = GSM. 2) if I do show sip register in asterisk CLI, I can see I'm registered (or I may be misinterpretting this command. What can I do to investigate this registration message ? Is there an special debug command ? thanks :) From: Jean Marc Le Fevre [EMAIL PROTECTED] Date: Wed, 18 Apr 2007 18:14:41 +0200 Hello all, I'm having a quite simple configuration like: SIP provider = asterisk SIP = lan Everythings works fine but sometime I can't get incoming call. Define sometimes and from where the income call you can't get? here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance [good stuff sniffed] Where do you suspect the error message is? --- Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Does this message make sense, not registered? Yuan Liu Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net;tag=00-31057-001dc208-591e1ca81 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df66 Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net;tag=00-32700-001dc209-6fc2b3303 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' sip.conf [general] context=incoming realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all allow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register = 09:[EMAIL PROTECTED] registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=6 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXX username=09XXX dtmfmode=inband qualify=6 fromdomain=freephonie.net [freephonie_inbound] type=peer context=incoming host=freephonie.net qualify=6 allow=all deny=0.0.0.0/0.0.0.0 permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net etension.conf ... [incoming] exten = s,1,Ringing exten = s,2,Noop(I receive a sip call); exten = s,n,Goto(home,1000,1) exten = s,n,Congestion ; ... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:4627b6c350701639315548! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming SIP call
Hello and thanks for answering, As I just answer to Yuan LIU, what I don't understand, is that I can place an outbound call from asterisk to a gsm at the same time I can't get asterisk thought a inbound call. But I'll try what you advice me. I'll tell you the result of it Jean-Marc LE FEVRE Le 19 avr. 07 à 20:08, Bala Neelakantan a écrit : If your SIP server loses REGISTERs then it cant place an inbound SIP call. Try changing the REGISTER frequency to lower value. When you see incoming SIP call fail, you might want to check whether the REGISTERs are working. Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Jean Marc Le Fevre Sent: Wednesday, April 18, 2007 11:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming SIP call Hello all, I'm having a quite simple configuration like: SIP provider = asterisk SIP = lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch= z9hG4bK67c2df66;rport From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;b ranch=z9hG4bK253c1a3d;rport From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: 7263e88c20c9f3 [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net;tag=00-31057-001dc208-591e1ca81 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df66 Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asteris k sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net;tag=00-32700-001dc209-6fc2b3303 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' sip.conf [general] context=incoming realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all all ow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register = 09:[EMAIL PROTECTED] registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=6 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXX username=09XXX dtmfmode=inband qualify=6 fromdomain=freephonie.net [freep honie_inbound] type=peer context=incoming host=freephonie.net qualify=6 allow=all deny=0.0.0.0/0..0.0.0 permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net etension.conf ... [incoming] exten = s,1,Ringing exten = s,2,Noop(I receive a sip call); exten = s,n,Goto(home,1000,1) exten = s,n,Congestion ; ... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:4627b30550701698699180! !DSPAM:4627b7bb50703422486060! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming SIP call
Hello all, I'm having a quite simple configuration like: SIP provider = asterisk SIP = lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch=z9hG4bK67c2df66;rport From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch=z9hG4bK253c1a3d;rport From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net;tag=00-31057-001dc208-591e1ca81 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df66 Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net;tag=00-32700-001dc209-6fc2b3303 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' sip.conf [general] context=incoming realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all allow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register = 09:[EMAIL PROTECTED] registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=6 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXX username=09XXX dtmfmode=inband qualify=6 fromdomain=freephonie.net [freephonie_inbound] type=peer context=incoming host=freephonie.net qualify=6 allow=all deny=0.0.0.0/0.0.0.0 permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net etension.conf ... [incoming] exten = s,1,Ringing exten = s,2,Noop(I receive a sip call); exten = s,n,Goto(home,1000,1) exten = s,n,Congestion ; ... !DSPAM:462643f450705772331342! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail: How to send a notification even if Caller does not let any messages?
Hi, First, sorry to repost, As I didn't get any replies, maybe this time, I will get more lucky. I was wondering if there was a way in Asterisk (agi script, asterisk-itself, whatever ... ) to send a notification to the user (Mail, SMS like voicemail application is doing) if the user has called, but did not leave any messages? I tried to use the minmessage, but, couldn't. Is that the way ? I was thinking of using the h Dialplan, and launch some script, but then, how to know if caller has left a message or not ? I wouldn't like to send 2 messages to the user. Thanks for your help ! Jean-Marc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
