Re: [asterisk-users] Question on resources
Hi, Am Donnerstag, dem 04.08.2022 um 20:32 -0400 schrieb Jerry Geis: > I am running Asterisk 13.30.0 > 40 core CPU (VM) VMware. > CentOS 7 > 32 G ram > 10G vmx network > > Should be plenty of room for anything... > > Yes asterisk is running 270% CPU... > Is it not taking advantage of the 40 cores ? > I am bring around 300 SIP endpoints in a muted audio conference (so > one way) and this spikes up the CPU to 270%. What type of conference? Is it meetme or confbridge? AFAIK meetme is working on a single thread... > > Is there something I dont have set right to take advantage to > the resourses? > Thanks > > Jerry HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pickup with pjsip not working
Am Dienstag, dem 01.03.2022 um 07:46 -0500 schrieb Tom Ray: > What phone is this being used on? I am able to do call pickup on > various Polycom VVX’s, Yealink’s and even old Cisco SPA3xx/5xx > phones. I think I even got it on an old snom but I would have to fire > it up to double check. > > Granted that each the phone configs for each of those brands do ask > for a pickup code (some have default codes). So knowing what phones > you are trying to do this with might help solve it. > > > Tom I found it on Snom phones with current firmware and also on Yealink phones. But it may be configurable on these phones. I'll check this first. Karsten > > From: asterisk-users On > Behalf Of Joshua C. Colp > Sent: Tuesday, March 1, 2022 6:36 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > Subject: Re: [asterisk-users] Pickup with pjsip not working > > On Tue, Mar 1, 2022 at 7:16 AM Karsten Wemheuer wrote: > > Am Dienstag, dem 01.03.2022 um 06:37 -0400 schrieb Joshua C. Colp: > > > On Tue, Mar 1, 2022 at 6:14 AM Karsten Wemheuer > > wrote: > > > > Hi *, > > > > > > > > i am currently trying to migrate from chan_sip to pjsip. I am > > using > > > > Asterisk version 18.10. > > > > > > > > In chan_sip information about the pickup was sent in the XML > > body > > > > of > > > > the NOTIFY requests: > > > > > > > > /--- > > > > > > > > > version="3" > > > > state="full" > > > > entity="sip:213@192.168.10.70"> > > > > > > > direction="recipient"> > > > > > > > > > > > > \--- > > > > > > > > > > > > If I use pjsip, the pickup information is missing: > > > > > > > > /--- > > > > > > > > > version="1" > > > > state="full" entity="sip:213@192.168.10.75:25060"> > > > > > > > > > > > > > > > > \--- > > > > > > > > Many phones expect this information and cannot perform a > > pickup. > > > > Where does this need to be configured or does this not work in > > > > pjsip? > > > > > > It does not appear as though anyone has written support for this > > in > > > PJSIP. > > > > > Do You know, if someone is working on this? Maybe I can help. Is it > > part of the upstream project or would it be built somewhere into > > res/res_pjsip.XXX? > > > I know of noone working on this, and it would be part of Asterisk > itself. > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pickup with pjsip not working
Am Dienstag, dem 01.03.2022 um 06:37 -0400 schrieb Joshua C. Colp: > On Tue, Mar 1, 2022 at 6:14 AM Karsten Wemheuer wrote: > > Hi *, > > > > i am currently trying to migrate from chan_sip to pjsip. I am using > > Asterisk version 18.10. > > > > In chan_sip information about the pickup was sent in the XML body > > of > > the NOTIFY requests: > > > > /--- > > > > > state="full" > > entity="sip:213@192.168.10.70"> > > > direction="recipient"> > > > > > > \--- > > > > > > If I use pjsip, the pickup information is missing: > > > > /--- > > > > > state="full" entity="sip:213@192.168.10.75:25060"> > > > > > > > > \--- > > > > Many phones expect this information and cannot perform a pickup. > > > > Where does this need to be configured or does this not work in > > pjsip? > > It does not appear as though anyone has written support for this in > PJSIP. > Do You know, if someone is working on this? Maybe I can help. Is it part of the upstream project or would it be built somewhere into res/res_pjsip.XXX? Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pickup with pjsip not working
Hi *, i am currently trying to migrate from chan_sip to pjsip. I am using Asterisk version 18.10. In chan_sip information about the pickup was sent in the XML body of the NOTIFY requests: /--- \--- If I use pjsip, the pickup information is missing: /--- \--- Many phones expect this information and cannot perform a pickup. Where does this need to be configured or does this not work in pjsip? Thanks for any help Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Hi Luca, Am Samstag, den 13.06.2020, 08:28 +0200 schrieb Luca Bertoncello: > Hi! > > I have a Asterisk installation to manage my phones at home (provider > is > Deutsche Telekom). > It works, but very often the voice is "broken"... > Yesterday during a call it was very difficult to understand what my > partner sayd... > > It can NOT be a problem of other downloads/uploads, since in that > moment > there were no ones... The product is "All-IP" and not the SIP trunk, right? The call starts normally and after about 15 minutes the quality is disturbed? Try to set "session-timers = refuse" in the sip.conf in the global section. I have observed that when updating the session this error occurs. Best regards, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP TLS not working, Asterisk 16.9.0
Hi Stefan, thanks a lot. It is working now. Best regards, Karsten Am Freitag, den 01.05.2020, 18:40 +0200 schrieb Stefan Tichy: > Hi Karsten, > > > On Thu, Apr 30, 2020 at 05:50:39PM +0200, Karsten Wemheuer wrote: > > > > The server sends Server Hello, Certificate, Server Key > > Exchange and Server Hello Done. > Something in that packet seems to be unacceptable for openssl 1.1.1d > as it is compiled and configured for Buster. > > Certificate length, Digest algorithm, ... > > > You my change the system default settings at the bottom of > "/etc/ssl/openssl.cnf", restart asterisk and try again. Keep in > mind that this will affect the whole server. > > > > > -- > Stefan Tichy ( asterisk3 at pi4tel dot de ) > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP TLS not working, Asterisk 16.9.0
Hi, I have problems with SIP via TLS. Asterisk works as a client. The TCP connection is established, followed by a client hello from Asterisk to the server. The server sends Server Hello, Certificate, Server Key Exchange and Server Hello Done. Than Asterisk sends back a Alert (Level: Fatal, Description Handshake Failure). The following line appears in the log: ast_iostream_start_tls: Problem setting up ssl connection: error:0001:lib(0):func(0):reason(1), Internal SSL error Asterisk version is 16.9.0, openssl is 1.1.1d-0+deb10u2 of debian Buster. The configuration works with Asterisk 11.25 and openssl 1.0.1. Any hints on how to find the error? Best regards, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] internal call record
Hi,
Am Sonntag, den 10.03.2019, 12:46 +0300 schrieb Gokan Atmaca:
> Hello
>
> Mynum: 6001 , Othernum: 6002.
>
>
> I can record as follows. But I do not enter individual records for
> each internal
> required. I want to do it more smoothly with a Macro.
>
> Thanks.
>
> exten => _6001,1,NoOp()
> exten => _6001,n,MixMonitor(${UNIQUEID}.wav,ab)
> exten => _6001,n,Dial(SIP/6001,20)
> exten => _6001,n,StopMixMonitor()
> exten => _6001,n,Hangup()
>
If You are using SIP, pay attention to media setup (option
"directmedia" in case of chan_sip).
Using directmedia the media flows from end to end not running through
asterisk. In this case recording doesn't work.
HTH,
Karsten
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[asterisk-users] Missing audio on playback in 16.0
Hi, I am currently evaluating asterisk 16. I have noticed an issue using application playback. The beginning and the end of the audio file are missing. If I use answer and wait(1) before playback, the beginning is correct. I am using chan_sip, if this is of interest. Best regards Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes
Hi Luca,
Am Montag, den 21.12.2015, 18:52 +0100 schrieb Luca Bertoncello:
> Hi list!
>
> My Problem: all calls to international numbers will be dropped after exactly
> 15 minutes...
> I have a VoIP-account by Deutsche Telekom.
> This is what I see when I call someone (my parents) and the connection will
> be dropped:
>
> == Using SIP RTP CoS mark 5
> -- Executing [+3901522@default:1] Set("SIP/004935-0125",
> "newNumber=003901522") in new stack
> -- Executing [+3901522@default:2]
> Verbose("SIP/004935-0125", "2,Rewrite number +3901522 to
> 003901522") in new stack
> == Rewrite number +3901522 to 003901522
> -- Executing [+3901522@default:3] Dial("SIP/004935-0125",
> "local/003901522") in new stack
> -- Called local/003901522
> -- Executing [003901522@default:1]
> Verbose("Local/003901522@default-003c;2", "2,DEFAULT") in new stack
> == DEFAULT
> -- Executing [003901522@default:2]
> Set("Local/003901522@default-003c;2", "CHANNEL(musicclass)=default")
> in new stack
> -- Executing [003901522@default:3]
> GotoIf("Local/003901522@default-003c;2", "0?dialrebvoice") in new
> stack
> -- Executing [003901522@default:4]
> GotoIf("Local/003901522@default-003c;2", "0?dialluca") in new stack
> -- Executing [003901522@default:5]
> GotoIf("Local/003901522@default-003c;2", "1?dialluca") in new stack
> -- Goto (default,003901522,13)
> -- Executing [003901522@default:13]
> Verbose("Local/003901522@default-003c;2", "2,Outgoing call for
> 003901522 using pbxluca") in new stack
> == Outgoing call for 003901522 using pbxluca
> -- Executing [003901522@default:14]
> Dial("Local/003901522@default-003c;2",
> "SIP/pbxluca/003901522,,RXx") in new stack
> == Using SIP RTP CoS mark 5
> -- Called SIP/pbxluca/003901522
> -- SIP/pbxluca-0126 is ringing
> -- SIP/pbxluca-0126 is making progress passing it to
> Local/003901522@default-003c;2
> -- Local/003901522@default-003c;1 is ringing
> -- Local/003901522@default-003c;1 is making progress passing it
> to SIP/004935-0125
> -- SIP/pbxluca-0126 answered Local/003901522@default-003c;2
> -- Local/003901522@default-003c;1 answered
> SIP/004935-0125
> == Spawn extension (default, 003901522, 14) exited non-zero on
> 'Local/003901522@default-003c;2'
> -- fixed jitterbuffer created on channel SIP/004935-0125
> == Spawn extension (default, +3901522, 3) exited non-zero on
> 'SIP/004935-0125'
> -- fixed jitterbuffer destroyed on channel SIP/004935-0125
>
> My number is the 004935 and I called the 003901522.
> Any idea?
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
the timeout value of 15 minutes directs me to an issue with session
timer. Try to refuse them by putting the line
session-timers = refuse
into the general context of sip.conf. Reload the sip stack with "sip
reload".
(I assume You are using chan_sip. I don't know how to disable session
timer in pj sip).
HTH,
Karsten
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Re: [asterisk-users] Semi OT - LDAP multi-valued attributes support in SIP phones
Hi Olivier, Am Donnerstag, den 29.01.2015, 18:07 +0100 schrieb Olivier: Hello, I've just started to look at LDAP in IP telephony. 1. I've read parts of RFC2798 which defines inetOrgPerson class. I could find homePhone or telephoneNumber (multi-valued) attributes but nothing like phoneExtension. Did I miss something ? I not, what would you advise to store private extensions in LDAP db ? 2. Several SIP phones can query LDAP databases. Have you met any success with such SIP phones and LDAP entries with two values for telephoneNumber ? In other words, is it required by SIP phone to only have one telephoneNumber or mobile attribute value per entry ? it is some time ago that have looked at this kind of stuff... AFAIK most phones have the possibility to define some sort of filters to select the attribute(s). If I remember correctly the phones from Snom, Gigaset, Tiptel and Yealink have a parameter to select attributes for name and number. HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk sends CANCEL to the wrong destination
Hi, Am Dienstag, den 16.12.2014, 16:32 +0100 schrieb Karsten Wemheuer: Hi, I got a weird behaviour in asterisk (original found in 1.8 but it is still the same in 11.15.0). I have three phones communicating via OpenSIPs with asterisk. Phone A dials 100 and asterisk calls SIP/phone-b. Phone B accepts the call. The User on Phone B places the call on hold, dials 200 and, while hearing the dial tone of ringing Phone C, places the handset on hook. Phone B sends a REFER, so that Phone A is connected with the ringing Phone C. Asterisk sends an UPDATE to Phone-C to update the connected line information. Now the user on Phone B realized that User B is not available. He presses the blinking LED of BLF to get the Call back. A and B are now connected again. But Phone C is still ringing. Asterisk sends the CANCEL to terminate the call to phone C not to the proxy (where the INVITE comes from), but directly to the phone. The phone ignores this CANCEL, as it does not belong to a call and so the phone keeps on ringing. If I modify the configuration, so that there is no UPDATE for the connected line information, the CANCEL is send via proxy to the phone and all is well. I possibly find the source of the failure and a patch, working without failure in my test scenario. I filed a bug with patch attached, see https://issues.asterisk.org/jira/browse/ASTERISK-24628 Have a nice day, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI Redirect both calls from a bridge
Hi Neil, Am Mittwoch, den 17.12.2014, 09:08 -0500 schrieb Neil Cherry: Doe anybody know of a way to redirect both channels from a bridge to different dial plan extensions from the using the AMI. Currently, as soon as I redirect one of the channels the other appears to be dropped and gets reorder tone (congestion, fast busy). I guess what I really need is a way to redirect one of the channels and hold on to the other. I think You have to do it in two steps. First connect both legs with a conference and then connect each one with the final extension. You didn't tell, which version of asterisk You are using. In 11 and later there is the new conference module, which makes it easier. In the first step You can use AMI REDIRECT to transfer both parties into one dynamic conference. Use the Channel and ExtraChannel to take both channels. In the second step use AMI Join Events to trigger your next transfer to the different extensions in Your dialplan. Each channel joining the conference will generate a separate event. HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk sends CANCEL to the wrong destination
Hi, I got a weird behaviour in asterisk (original found in 1.8 but it is still the same in 11.15.0). I have three phones communicating via OpenSIPs with asterisk. Phone A dials 100 and asterisk calls SIP/phone-b. Phone B accepts the call. The User on Phone B places the call on hold, dials 200 and, while hearing the dial tone of ringing Phone C, places the handset on hook. Phone B sends a REFER, so that Phone A is connected with the ringing Phone C. Asterisk sends an UPDATE to Phone-C to update the connected line information. Now the user on Phone B realized that User B is not available. He presses the blinking LED of BLF to get the Call back. A and B are now connected again. But Phone C is still ringing. Asterisk sends the CANCEL to terminate the call to phone C not to the proxy (where the INVITE comes from), but directly to the phone. The phone ignores this CANCEL, as it does not belong to a call and so the phone keeps on ringing. If I modify the configuration, so that there is no UPDATE for the connected line information, the CANCEL is send via proxy to the phone and all is well. In the following trace asterisk is at 192.168.10.70:25060, the proxy is at 192.168.10.5060. The phones are at 192.168.10.201 (Phone-A), 192.168.10.124 (Phone-B) and 192.168.10.102 (Phone-C). Shortened SIP-Trace is as follows (starting with the REFER): Call between A and B is on hold, B has call to C in ringing state. Transfer, so that A has the ringing call to C REFER is send from Phone B to proxy: U 192.168.10.124:5060 - 192.168.10.75:5060 REFER sip:180@192.168.10.75:25060 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.124:5060;branch=z9hG4bK_7C2F8008F5C1_T5B47A547;rport. From: sip:phone-b@192.168.10.75;tag=7C2F8008F5C1_T1484940980. To: PhoneA sip:180@192.168.10.75:25060;tag=as18f69e58. Call-ID: 452af6610540b7cf0d4c49f372d46779@192.168.10.75. CSeq: 2 REFER. Refer-To: sip:200@192.168.10.75?Replaces=CALL_ID9_7C2F8008F5C1_T1871060912% 40192_168_10_124%3Bto-tag%3Das13cb6557%3Bfrom-tag% 3D7C2F8008F5C1_T570909484. Referred-by: sip:phone-b@192.168.10.75. Contact: sip:phone-b@192.168.10.124:5060. Max-Forwards: 70. Content-Length: 0. REFER is send from proxy to asterisk: U 192.168.10.75:5060 - 192.168.10.75:25060 REFER sip:180@192.168.10.75:25060 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T5B47A547. Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T5B47A547;rport=5060. From: sip:phone-b@192.168.10.75;tag=7C2F8008F5C1_T1484940980. To: PhoneA sip:180@192.168.10.75:25060;tag=as18f69e58. Call-ID: 452af6610540b7cf0d4c49f372d46779@192.168.10.75. CSeq: 2 REFER. Refer-To: sip:200@192.168.10.75?Replaces=CALL_ID9_7C2F8008F5C1_T1871060912% 40192_168_10_124%3Bto-tag%3Das13cb6557%3Bfrom-tag% 3D7C2F8008F5C1_T570909484. Referred-by: sip:phone-b@192.168.10.75. Contact: sip:phone-b@192.168.10.124:5060. Max-Forwards: 69. Content-Length: 0. U 192.168.10.75:25060 - 192.168.10.75:5060 SIP/2.0 202 Accepted. Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T5B47A547;received=192.168.10.75. Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T5B47A547;rport=5060. From: sip:phone-b@192.168.10.75;tag=7C2F8008F5C1_T1484940980. To: PhoneA sip:180@192.168.10.75:25060;tag=as18f69e58. Call-ID: 452af6610540b7cf0d4c49f372d46779@192.168.10.75. CSeq: 2 REFER. Server: IPTAM PBX (Version 20141216/6814). Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:180@192.168.10.75:25060. Content-Length: 0. Update connected line info on ringing phone C. Request is sent directly from asterisk to phone: U 192.168.10.75:25060 - 192.168.10.102:2048 UPDATE sip:phone-c@192.168.10.102:2048;line=jrtagx14 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK7bc63ea0. Max-Forwards: 70. From: PhoneB sip:100@192.168.10.75:25060;tag=as2b910e05. To: sip:phone-c@192.168.10.75;tag=dg06weh6iz. Contact: sip:100@192.168.10.75:25060. Call-ID: 2dadc6b61efcb4dc726be746564897bf@192.168.10.75. CSeq: 103 UPDATE. User-Agent: IPTAM PBX (Version 20141216/6814). P-Asserted-Identity: PhoneA sip:180@192.168.10.75. X-Asterisk-rpid-update: Yes. Content-Length: 0. U 192.168.10.75:5060 - 192.168.10.124:5060 SIP/2.0 202 Accepted. Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T5B47A547;rport=5060. From: sip:phone-b@192.168.10.75;tag=7C2F8008F5C1_T1484940980. To: PhoneA sip:180@192.168.10.75:25060;tag=as18f69e58. Call-ID: 452af6610540b7cf0d4c49f372d46779@192.168.10.75. CSeq: 2 REFER. Server: IPTAM PBX (Version 20141216/6814). Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:180@192.168.10.75:25060. Content-Length: 0. U 192.168.10.102:2048 - 192.168.10.75:25060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK7bc63ea0. From: PhoneB
Re: [asterisk-users] IAXModem or T38Modem?
Hi Mike, Am Montag, den 24.03.2014, 01:41 -0400 schrieb Mike Diehl: Hi all, I'm installing Hylafax on my Asterisk system. From what I've read, I can either use IAXModem or T38Modem to provide the virtual fax device. So at the risk of starting a religious war, which one should I use? I don't mind running IAX if I have to. I want as much flexibility and stability as I can get. So, what are your recommendations? It depends on Your environment and Your asterisk version. If Your connection to the PSTN is via ISDN (eg. channel via DAHDI or CAPI), You should use IAXmodem. The fax is transferred on the audio layer and there is no need to translate it into T.38. If Your connection is via some VoIP Provider using T.38 or via a Mediagateway like BeroFix, You should use T38Modem. Newer versions of asterisk have additional features regarding T.38 (AFAIK). In this case there may be no need for any of the modems. HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to configure asterisk to only accept SIP from kamailio@localhost but exchange RTP on all interfaces?
Hi Alex, Am Dienstag, den 25.02.2014, 13:04 -0500 schrieb Alex Villacís Lasso: El 25/02/14 08:30, Karsten Wemheuer escribió: Hi Alex, Am Donnerstag, den 20.02.2014, 13:48 -0500 schrieb Alex Villacís Lasso: I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration (MySQL database) so that kamailio authenticates and then forwards the registration to asterisk on localhost. The setup calls for asterisk to be configured to listen for SIP traffic on all interfaces, on a nonstandard port (I chose 5080). It also calls for blanking of the password for the SIP peer (in my case, a softphone), so that it will not request for authentication again. I have managed to make a call with working audio from the softphone to an extension on asterisk through kamailio. My concern is that asterisk is left listening for SIP through all interfaces and with no SIP passwords. I want to secure the setup against directed traffic to the asterisk UDP port (5080), that bypasses the kamailio process. I tried setting bindaddr=127.0.0.1 so asterisk will only listen for SIP traffic on localhost, but this has the side effect of also removing audio - the call appears to be successful on the softphone and on the asterisk logs, but no audio is actually heard. My theory is that the RTP traffic is being sent to kamailio instead of the softphone. How can I set up asterisk so that it can send RTP anywhere but reject any SIP traffic that does not come from the kamailio process on localhost? If You bind asterisk to 127.0.0.1 I think the media connection is set for this IP. Your Softphone can not reach the correct 127.0.0.1 (localhost is everywhere). I would suggest, You setup asterisk on eth0 address or 0.0.0.0. In the sip.conf You could secure Your setup with deny = 0.0.0.0/0.0.0.0 permit = Your-LAN-Adress This way asterisk accepts SIP from Your box only. This might work, but would need to touch sip.conf every time the IP address changes. It would be nice to have a configuration that can be set up once and not modified again. That is why I wanted to set up localhost. It is the LAN address of Your Server, where asterisk and kamailio are running. The permit entry allows communication between kamailio and asterisk. Why would You change this address? Maybe I don't understand Your setup. Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to configure asterisk to only accept SIP from kamailio@localhost but exchange RTP on all interfaces?
Hi Alex, Am Donnerstag, den 20.02.2014, 13:48 -0500 schrieb Alex Villacís Lasso: I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration (MySQL database) so that kamailio authenticates and then forwards the registration to asterisk on localhost. The setup calls for asterisk to be configured to listen for SIP traffic on all interfaces, on a nonstandard port (I chose 5080). It also calls for blanking of the password for the SIP peer (in my case, a softphone), so that it will not request for authentication again. I have managed to make a call with working audio from the softphone to an extension on asterisk through kamailio. My concern is that asterisk is left listening for SIP through all interfaces and with no SIP passwords. I want to secure the setup against directed traffic to the asterisk UDP port (5080), that bypasses the kamailio process. I tried setting bindaddr=127.0.0.1 so asterisk will only listen for SIP traffic on localhost, but this has the side effect of also removing audio - the call appears to be successful on the softphone and on the asterisk logs, but no audio is actually heard. My theory is that the RTP traffic is being sent to kamailio instead of the softphone. How can I set up asterisk so that it can send RTP anywhere but reject any SIP traffic that does not come from the kamailio process on localhost? If You bind asterisk to 127.0.0.1 I think the media connection is set for this IP. Your Softphone can not reach the correct 127.0.0.1 (localhost is everywhere). I would suggest, You setup asterisk on eth0 address or 0.0.0.0. In the sip.conf You could secure Your setup with deny = 0.0.0.0/0.0.0.0 permit = Your-LAN-Adress This way asterisk accepts SIP from Your box only. HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sipgate outgoing calls
Hi, Am Mittwoch, den 18.09.2013, 14:29 +0100 schrieb gpxctawjc...@irational.org: Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line What Sipgate product are You using? At least in Germany there are different configurations for the different products necessary. For Sipgate trunking and Sipgate team You have to configure an outboundproxy (which differs between both products). For Sipgate Basic you don't need an outboundproxy. As far as I remember there was an issue with some asterisk versions and the outboundproxy for Sipdate team. Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error 488 Not Acceptable Here
Hi, Am Donnerstag, den 23.05.2013, 20:48 +0200 schrieb Maximilian Grobecker: Am 22.05.2013 16:39, schrieb Andrew Colin: Hi guys, Any idea why I am getting this error when someone tries to send me a T38 Fax? Hi, Maybe you have not allowed T.38 as acceptable codec ;-) You can try with allow=all in your sip.conf. No, T.38 is not a codec and so allow=all will not help. To use T.38 You have to enable T.38 with t38pt_udptl = yes in sip.conf. The reason, why You get a 488 Not Acceptable Here488 Not Acceptable Here, is only detectable with a SIP Trace. There are many reasons e.g. - Your asterisk version does not support T.38 - T.38 is not enabled (see above) - T.38 is enabled, but not at the relevant peers (in most versions of asterisk there is only support for T.38 passthrough, so both peers have to support T.38) - There are problems in the transmission and some peers wants to switch back to audio level and the other or asterisk is not willing to support this. The last reason may occur, if You have NAT and do not correctly forward the data ports of T.38 (UDPTL Ports). Best way is to get a SIP Trace to analyse. If You provide one, You should also tell, which version of asterisk. HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and hylafax: how to debug ...
Hi, Am Dienstag, den 07.05.2013, 21:48 +0200 schrieb Sebastian Niehaus: Am 07.05.2013 18:23, schrieb Sebastian Niehaus: For some reason, t38modem tells hylafax the line is BUSY so there is no fax send. Well, I may add the log of t38modem (sorry for the ugly formatting) Parts I consider as most important are: ModemConnection::SetUpConnection dstNum=189659 srcNum=30 srcName=root ... denied (all modems busy) [ snip ] it seems to me, that the call is routed from the modem to the modem (and not to asterisk). t38modem has some config options for call routing. Something like: route=modem:.*=sip:dn@ip:port route=sip:.*=modem:dn The first rule routes calls from the modem to a sip destination. I think in Your setup it should be route=modem:.*=sip:dn@127.0.0.1:5060. (I never used localhost in a setup like this, it should work with the IP of Your ethernet too). HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID external call after Attended Transfer
Hi, Am Montag, den 04.02.2013, 14:45 +0100 schrieb Jonas Kellens: Hello, thanks you for your answer. The IP-phones in this case are Yealink T32G. What setting is needed in this IP-phone ? as Kevin already written, set this in asterisk: sendrpid=pai trustrpid=yes I don't know the T32G, but in T2x series there is a setting under Accounts-Advanced called Caller ID Header. Select PAI or PAI +FROM. The default is FROM which won't work. HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail not working with vm boxes named with a star
Hi list, in asterisk 1.4 and maybe earlier it was possible to use voicemail system with mailboxes starting with some special characters like *. The line in voicemail.conf was like this: *123 = , AB,,,tz=cet|attach=no| Calling exten = s,n,Voicemail(*123,su) is working in asterisk 1.4. In Asterisk 1.8 the above scenario is not working any more. The Voicemail application reports an error message: WARNING: app_voicemail.c: leave_voicemail: No entry in voicemail config file for '*123' Is this a known bug, fixed in newer versions (I currently use 1.8.11) or should I file a bug report? Thanks, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail not working with vm boxes named with a star
Hi list,
Am Donnerstag, den 20.09.2012, 09:28 +0200 schrieb Karsten Wemheuer:
Hi list,
in asterisk 1.4 and maybe earlier it was possible to use voicemail
system with mailboxes starting with some special characters like *. The
line in voicemail.conf was like this:
*123 = , AB,,,tz=cet|attach=no|
Calling exten = s,n,Voicemail(*123,su) is working in asterisk 1.4.
In Asterisk 1.8 the above scenario is not working any more. The
Voicemail application reports an error message:
WARNING: app_voicemail.c: leave_voicemail: No entry in voicemail config
file for '*123'
Is this a known bug, fixed in newer versions (I currently use 1.8.11) or
should I file a bug report?
After looking at log files and source code, I found out, that in
function find_or_create in app_voicemail.c there is a statement:
if (!ast_strlen_zero(box) box[0] == '*') {
right at the beginning of that function. This leads to not setting up a
mailbox *123, whereas 123* is allowed. The logging says:
The '*' character, when it is the first character in a mailbox
or password, is used to jump to a predefined extension 'a'. A
mailbox or password beginning with '*' is not valid and will be
ignored.
I do not see, why a mailbox should not be valid starting with '*'. The
feature to jump to a predefined extension by pressing * exists in
Asterisk 1.4 but you can create a mailbox starting with '*' in 1.4. If
for some reason the feature to jump to an predefined extension by
pressing some key forbids using that key as first part of a mailbox,
than the above code should prevent using '0' as first part too, I think.
Pressing '0' is analog to pressing '*' according to the documentation.
Does anybody know, why the if-statement is put in the code?
Thanks,
Karsten
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Re: [asterisk-users] Voicemail not working with vm boxes named with a star
Hi Matthew, Am Donnerstag, den 20.09.2012, 06:27 -0500 schrieb Matthew Jordan: - Original Message - From: Karsten Wemheuer k...@gmx.de To: asterisk-users@lists.digium.com Sent: Thursday, September 20, 2012 2:28:07 AM Subject: [asterisk-users] Voicemail not working with vm boxes named with a star Hi list, in asterisk 1.4 and maybe earlier it was possible to use voicemail system with mailboxes starting with some special characters like *. The line in voicemail.conf was like this: *123 = , AB,,,tz=cet|attach=no| Calling exten = s,n,Voicemail(*123,su) is working in asterisk 1.4. In Asterisk 1.8 the above scenario is not working any more. The Voicemail application reports an error message: WARNING: app_voicemail.c: leave_voicemail: No entry in voicemail config file for '*123' Is this a known bug, fixed in newer versions (I currently use 1.8.11) or should I file a bug report? Nope, this is not a bug. The change in behavior was deliberate (see https://issues.asterisk.org/jira/browse/ASTERISK-17433). Starting a mailbox with a '*' conflicted with the auto-attendant feature in app_voicemail, wherein a user can be redirected to the 'a' extension by sending the '*' DTMF. There were a number of weird side effects that occurred due to this, most of which involved users who had created a mailbox that began with a '*' unable to check their voicemail. Since it would be very easy to mis-configure your system if you had both the auto-attendant feature enabled and allowed users to have a mailbox/password that started with '*', we prevented the latter scenario. Thank You for Your response. I got Your message after writing my second message. I understand the problem. But if the issue is triggered mixing boxes starting with * and the assistant feature, what about the operator feature? Should boxes starting with '0' also be prevented? Thanks, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.10
Hii Am Montag, den 11.06.2012, 16:12 -0700 schrieb motty.cruz: Hello, How to change ring tone on interncal call? I'm using Centos 5.8 Asterisk 1.8 exten = 666,1,SIPAddHeader(Alert-Info:http://1.2.3.4/ringtones/ghost.wav) exten = 666,n,Dial(SIP/10) The above would not how to defirenciate from internal call or external call? Warren told one way to differentiate between internal and external call. Another would be to set a variable in incoming context and check against it before set up the additional header. Take into account, that this feature is vendor specific as Yaroslav said. To my knowledge snom, Tiptel, Yealink supports ring tones via Alert-Info. Aastra phones supports it too, but You can only select between bellcore cadences (no download of custom ringtones). Snom phones needs sln format whereas tiptel/yealink needs ulaw coding. Other phones may support this feature too (Grandstream?, Linksys?, Polycom?). HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 Transfer CallerID
Hi, Am Dienstag, den 08.05.2012, 14:13 +0200 schrieb Jonas Kellens: Hello, when a call comes in and is answered by colleague A, this colleague A sees the CallerID of the external calling number. When colleague A transfers the call to colleague B, attended or unattended, then colleague B sees the number of colleague A on his screen while talking to the external calling number. I expect here that colleague B would see the external calling number on the screen of his IP-phone. How can I get this behaviour ? As far as I understand You, you want to update the callerID (A calls B, the call is established, then A transfers the external party to B). In this case (and if You are using SIP endpoints), check out the config params sendrpid and rpid_update in sip.conf. AFAIK this feature is working in versions from 1.8 and newer. HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDISendCallreroutingFacility
Hi, Am Samstag, den 10.03.2012, 08:42 -0800 schrieb Mehdi Shirazi: Hi I installed Asterisk 1.8.7 with CD ISO(Elastix 2.2) I want to use DAHDISendCallreroutingFacility Application on a PRI link(LIBPRI Already installed). according to https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 Asterisk 1.8 include this application but I cannot see it with core show applications Do I need to install mISDN or other modules for using that ? Regards M.Shirazi No You don't need mISDN or other modules. DAHDISendCallreroutingFacility is part of chan_dahdi. But as far as I know Asterisk 1.8.7 had problems with this application. Try using at least 1.8.8 (1.8.10.0 is currently the stable version of 1.8 release). HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem while sending SIP NOTIFY via AMI in 1.8.10-rc2
Hi, while testing asterisk 1.8.10-rc2 I stumbled across a weird behavior. I want to notify a snom phone to reload its configuration. For this to happen, I use the NOTIFY mechanism. I started the notify via AMI command. Asterisk is bound to udp 25060, because all phones are registered with a local opensips proxy which uses 5060. The expected behavior would be: asterisk send SIP NOTIFY to the proxy, the proxy sends it to the phone. Actually asterisk sends the packet to the proxy, but the contact header contains something invalid (IMHO): On Manager Interface: T 127.0.0.1:57530 - 127.0.0.1:5038 [AP] Action: SIPnotify. Channel: SIP/max. Variable: Event=check-sync\;reboot=false. Leads to: U 192.168.10.72:25060 - 192.168.10.72:5060 NOTIFY sip:max@192.168.10.72 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.72:0;branch=z9hG4bK1dff6efe. Max-Forwards: 70. From: asterisk sip:asterisk@192.168.10.72;tag=as66766c2a. To: sip:max@192.168.10.72. Contact: sip:asterisk@192.168.10.72:0. Call-ID: 412a8eff76bd7ac56ac06831256fd6aa@192.168.10.72. CSeq: 102 NOTIFY. Subscription-State: terminated. Event: check-sync;reboot=false. Content-Length: 0. The weird thing is the port number 0 in the contact header. Is this a bug or do I something wrong? Thanks, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem while sending SIP NOTIFY via AMI in 1.8.10-rc2
Hi, a little extension to my previous post: The phone sends 200 OK for the NOTIFY via proxy to asterisk, but asterisk seems to ignore this. About 500 ms later, the NOTIFY is repeated by asterisk. This continues up to the final timeout (with the typical log message). Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2
Hi,
I got a problem with asterisk 1.8.9.2. The same scenario is working fine
in 1.8.8.2.
Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on
the same LAN, no NAT.
Asterisk sends the INVITE to the proxy, the proxy sends INVITE to the
phone. The phone sends 180 RINGING back to the proxy. The proxy sends
180 RINGING to asterisk. So far so good. If the calling side decides to
cancel the call, asterisk sends the CANCEL directly to the phone. The
phone doesn't find the call and answers 404. In asterisk 1.8.8.2
asterisk sends the CANCEL to the proxy, which sends the CANCEL to the
phone and all ist fine.
I think, the new behavior comes from the lines
parse_ok_contact(p, req);
if (!reinvite) {
build_route(p, req, 1);
}
which are inserted in the handling of provisional SIP response.
Am I doing something wrong or is this a bug?
Thanks,
Karsten
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Re: [asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2
Hi Kevin,
Am Dienstag, den 14.02.2012, 09:46 -0600 schrieb Kevin P. Fleming:
On 02/14/2012 09:30 AM, Karsten Wemheuer wrote:
Hi,
I got a problem with asterisk 1.8.9.2. The same scenario is working fine
in 1.8.8.2.
Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on
the same LAN, no NAT.
Asterisk sends the INVITE to the proxy, the proxy sends INVITE to the
phone. The phone sends 180 RINGING back to the proxy. The proxy sends
180 RINGING to asterisk. So far so good. If the calling side decides to
cancel the call, asterisk sends the CANCEL directly to the phone. The
phone doesn't find the call and answers 404. In asterisk 1.8.8.2
asterisk sends the CANCEL to the proxy, which sends the CANCEL to the
phone and all ist fine.
I think, the new behavior comes from the lines
parse_ok_contact(p, req);
if (!reinvite) {
build_route(p, req, 1);
}
which are inserted in the handling of provisional SIP response.
Am I doing something wrong or is this a bug?
It's impossible to answer that question without seeing the SIP
signaling. The answer will depend on what the proxy did to insert itself
in the path (or not) when it forwarded the 180 RINGING response to Asterisk.
I shorten the trace to (hopefully) the relevant things. Asterisk is on
192.168.10.72, port 25060, proxy is opnesips on the same machine with
port 5060, the phone which is ringing is on 192.168.10.221.
Asterisk = Proxy:
U 192.168.10.72:25060 - 192.168.10.72:5060
INVITE sip:arthur@192.168.10.72 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
Max-Forwards: 70.
From: Max M..ller sip:245@192.168.10.72;tag=as3cafd135.
To: sip:arthur@192.168.10.72.
Contact: sip:245@192.168.10.72:25060.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
... sdp cut of ...
Proxy = Asterisk
U 192.168.10.72:5060 - 192.168.10.72:25060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: Max M..ller sip:245@192.168.10.72;tag=as3cafd135.
To: sip:arthur@192.168.10.72.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
Proxy = phone
U 192.168.10.72:5060 - 192.168.10.221:34381
INVITE sip:arthur@192.168.10.221:34381;line=478vzxb3 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.72;branch=z9hG4bK24be.5163d992.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
Max-Forwards: 69.
From: Max M..ller sip:245@192.168.10.72;tag=as3cafd135.
To: sip:arthur@192.168.10.72.
Contact: sip:245@192.168.10.72:25060.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
... sdp cut of ...
Phone = Proxy
U 192.168.10.221:34381 - 192.168.10.72:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.10.72;branch=z9hG4bK24be.5163d992.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: Max M..ller sip:245@192.168.10.72;tag=as3cafd135.
To: sip:arthur@192.168.10.72;tag=cvovqkf6i5.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
Contact: sip:arthur@192.168.10.221:34381;line=478vzxb3;reg-id=1.
Proxy = Asterisk
U 192.168.10.72:5060 - 192.168.10.72:25060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: Max M..ller sip:245@192.168.10.72;tag=as3cafd135.
To: sip:arthur@192.168.10.72;tag=cvovqkf6i5.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
Contact: sip:arthur@192.168.10.221:34381;line=478vzxb3;reg-id=1.
When canceling the call, asterisk sends
Asterisk = Phone
U 192.168.10.72:25060 - 192.168.10.221:34381
CANCEL sip:arthur@192.168.10.72 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
Max-Forwards: 70.
From: Max M..ller sip:245@192.168.10.72;tag=as3cafd135.
To: sip:arthur@192.168.10.72.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 CANCEL.
The Phone responds:
U 192.168.10.221:34381 - 192.168.10.72:25060
SIP/2.0 404 Not found.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: Max M..ller sip:245@192.168.10.72;tag=as3cafd135.
To: sip:arthur@192.168.10.72.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 CANCEL.
As noted in the earlier mail, this scenario is working in previous
versions (1,4.x up to asterisk 1.8.8.2).
Do You have any idea where the failure happens? Is it the proxy
configuration or is it at the asterisk side (maybe config or bug)?
Thanks,
Karsten
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Re: [asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2
Hi, Am Dienstag, den 14.02.2012, 11:32 -0600 schrieb Kevin P. Fleming: This does appear to be a bug in Asterisk; please open an issue in JIRA, and post the issue number here, so we can get someone looking at this ASAP. Thanks! Done, issue ASTERISK-19358. If I can do anything to test something, let me know. Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Analoge and E1 ports
Hi Bilal, Am Sonntag, den 22.01.2012, 13:06 -0800 schrieb bilal ghayyad: Hi All; Is there a telephony card that contains analoge ports and E1s at the same time? Beronet in Germany produces modular media gateways as cards to plug in a pc (PCI or PCI express) or as an external box. Each gateway takes up to 2 modules. There are modules available with 1 or 2 E1 or 4 FXS or 4 FXO (and other interfaces like BRI, GSM too). So You could get a gateway with e.g. 1xE1 and 4xFXS or 2xE1 and 4FXS. http://www.beronet.com HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Option 'd' of application Dial not working in 1.8.8-rc2
Hi, in asterisk 1.8.7.0 option 'd' works as expected: Pressing a key while in ringing state puts the call to an one digit extension. In asterisk 1.8.8-rc2 this is not working anymore. After doing a diff on the code it seems to me, that in version 1.8.7 there is an autoanswer in application dial in case there is option 'd' present. Putting a Answer in the dialplan in front of the Dial-Statement solves the problem. Is this a bug or a feature? Thanks, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Option 'd' of application Dial not working in 1.8.8-rc2
Hi Richard, Am Mittwoch, den 02.11.2011, 09:26 -0500 schrieb Richard Mudgett: Hi, in asterisk 1.8.7.0 option 'd' works as expected: Pressing a key while in ringing state puts the call to an one digit extension. In asterisk 1.8.8-rc2 this is not working anymore. After doing a diff on the code it seems to me, that in version 1.8.7 there is an autoanswer in application dial in case there is option 'd' present. Putting a Answer in the dialplan in front of the Dial-Statement solves the problem. Is this a bug or a feature? Thanks, It was a necessary change. That automatic answer in the dial application broke DTMF attended transfer. See v1.8 SVN commit log -r336658. Thanks for the information. I'll fix it with an explicit Answer in front of the Dial application. Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with video phone call, error in sdp media handling?
Hi, I try to setup a video call and I sometimes get no video. I set up a Yealink VP 2009 and a Ninja Softphone. Both a in the same LAN. Asterisk release is 1.8.7.0. Call from Yealink to the Ninja is working fine, if I start the call in video mode. In this case I can switch between voice-only and video and back without any problem. If I try the opposite direction there is no video. The Ninja starts the call in voice-mode and try to add video in an second invite. The same happens, if I start the call in voice-mode from the Yealink phone. As far as I can see there seems to be something broken in SDP handling. In the following test phone1 is calling extension 200, which is extension of phone2. In case of failure phone1 sends: INVITE sip:200@192.168.10.75 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.106:5062;branch=z9hG4bK1784123944. From: Karsten sip:phone1@192.168.10.75;tag=1171101891. To: sip:200@192.168.10.75. Call-ID: 1555625029@192.168.10.106. CSeq: 1 INVITE. Contact: sip:phone1@192.168.10.106:5062. Content-Type: application/sdp. Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE. Max-Forwards: 70. User-Agent: VideoPhone-V8438 22.30.0.60 00:15:65:1b:20:3f. Supported: replaces,100rel. Allow-Events: talk,hold,conference,refer. Content-Length: 274. . v=0. o=- 20006 20006 IN IP4 192.168.10.106. s=SDP data. c=IN IP4 192.168.10.106. t=0 0. m=audio 10020 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. Asterisk sends to the second phone: INVITE sip:phone2@192.168.10.141:1116 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK3aa00bba. Max-Forwards: 70. From: User1 sip:phone1@192.168.10.75;tag=as6e33f30b. To: sip:phone2@192.168.10.141:1116. Contact: sip:phone1@192.168.10.75:5060. Call-ID: 73a216f9167396885e099d0f2e5d4ca2@192.168.10.75. CSeq: 102 INVITE. User-Agent: IPTAM PBX. Date: Wed, 19 Oct 2011 14:49:17 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. P-Asserted-Identity: User1 sip:phone1@192.168.10.75. Content-Type: application/sdp. Content-Length: 454. . v=0. o=root 1873948927 1873948927 IN IP4 192.168.10.75. s=Asterisk PBX 1.8.7.0-1. c=IN IP4 192.168.10.75. b=CT:384. t=0 0. m=audio 18858 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. m=video 16964 RTP/AVP 31 34 98 99 104. a=rtpmap:31 H261/9. a=rtpmap:34 H263/9. a=rtpmap:98 h263-1998/9. a=rtpmap:99 H264/9. a=rtpmap:104 MP4V-ES/9. a=sendrecv. So asterisks adds a second media attribute for video. The OK from the second phone looks like this: SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK3aa00bba. From: User1 sip:phone1@192.168.10.75;tag=as6e33f30b. To: sip:phone2@192.168.10.141:1116;tag=30873f0b0ea954d6. Call-ID: 73a216f9167396885e099d0f2e5d4ca2@192.168.10.75. CSeq: 102 INVITE. User-Agent: Ninja GlobalIPTel. Max-Forwards: 70. Contact: sip:phone2@192.168.10.141:1116. Content-Type: application/sdp. Content-Length: 322. . v=0. o=- 3528024652 3528024652 IN IP4 192.168.10.141. s=SIPCall. i=VoIPCall. c=IN IP4 192.168.10.141. t=0 0. m=audio 24608 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. m=video 24610 RTP/AVP 34. a=rtpmap:34 H263/9. a=sendrecv. There is also a m=video attribute. Asterisk sends the OK to the initiating device: SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.10.106:5062;branch=z9hG4bK1387721920;received=192.168.10.106. From: Karsten sip:phone1@192.168.10.75;tag=1171101891. To: sip:200@192.168.10.75;tag=as5d003051. Call-ID: 1555625029@192.168.10.106. CSeq: 2 INVITE. Server: IPTAM PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: sip:200@192.168.10.75:5060. Content-Type: application/sdp. Content-Length: 262. . v=0. o=root
Re: [asterisk-users] Problem with video phone call, error in sdp media handling?
Hi, thanks for Your quick response. But as You can see in the commented SIP-Messages, asterisk gets a voice call and sends out a INVITE with two media attributes for video and voice towards the destination. Karsten Am Mittwoch, den 19.10.2011, 10:40 -0500 schrieb Danny Nicholas: Just a WAG - if you start the call in voice-mode, the video codecs aren't loaded. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karsten Wemheuer Sent: Wednesday, October 19, 2011 10:37 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problem with video phone call, error in sdp media handling? Hi, I try to setup a video call and I sometimes get no video. I set up a Yealink VP 2009 and a Ninja Softphone. Both a in the same LAN. Asterisk release is 1.8.7.0. Call from Yealink to the Ninja is working fine, if I start the call in video mode. In this case I can switch between voice-only and video and back without any problem. If I try the opposite direction there is no video. The Ninja starts the call in voice-mode and try to add video in an second invite. The same happens, if I start the call in voice-mode from the Yealink phone. As far as I can see there seems to be something broken in SDP handling. In the following test phone1 is calling extension 200, which is extension of phone2. In case of failure phone1 sends: INVITE sip:200@192.168.10.75 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.106:5062;branch=z9hG4bK1784123944. From: Karsten sip:phone1@192.168.10.75;tag=1171101891. To: sip:200@192.168.10.75. Call-ID: 1555625029@192.168.10.106. CSeq: 1 INVITE. Contact: sip:phone1@192.168.10.106:5062. Content-Type: application/sdp. Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE. Max-Forwards: 70. User-Agent: VideoPhone-V8438 22.30.0.60 00:15:65:1b:20:3f. Supported: replaces,100rel. Allow-Events: talk,hold,conference,refer. Content-Length: 274. . v=0. o=- 20006 20006 IN IP4 192.168.10.106. s=SDP data. c=IN IP4 192.168.10.106. t=0 0. m=audio 10020 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. Asterisk sends to the second phone: INVITE sip:phone2@192.168.10.141:1116 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK3aa00bba. Max-Forwards: 70. From: User1 sip:phone1@192.168.10.75;tag=as6e33f30b. To: sip:phone2@192.168.10.141:1116. Contact: sip:phone1@192.168.10.75:5060. Call-ID: 73a216f9167396885e099d0f2e5d4ca2@192.168.10.75. CSeq: 102 INVITE. User-Agent: IPTAM PBX. Date: Wed, 19 Oct 2011 14:49:17 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. P-Asserted-Identity: User1 sip:phone1@192.168.10.75. Content-Type: application/sdp. Content-Length: 454. . v=0. o=root 1873948927 1873948927 IN IP4 192.168.10.75. s=Asterisk PBX 1.8.7.0-1. c=IN IP4 192.168.10.75. b=CT:384. t=0 0. m=audio 18858 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. m=video 16964 RTP/AVP 31 34 98 99 104. a=rtpmap:31 H261/9. a=rtpmap:34 H263/9. a=rtpmap:98 h263-1998/9. a=rtpmap:99 H264/9. a=rtpmap:104 MP4V-ES/9. a=sendrecv. So asterisks adds a second media attribute for video. The OK from the second phone looks like this: SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK3aa00bba. From: User1 sip:phone1@192.168.10.75;tag=as6e33f30b. To: sip:phone2@192.168.10.141:1116;tag=30873f0b0ea954d6. Call-ID: 73a216f9167396885e099d0f2e5d4ca2@192.168.10.75. CSeq: 102 INVITE. User-Agent: Ninja GlobalIPTel. Max-Forwards: 70. Contact: sip:phone2@192.168.10.141:1116. Content-Type: application/sdp. Content-Length: 322. . v=0. o=- 3528024652 3528024652 IN IP4 192.168.10.141. s=SIPCall. i=VoIPCall. c=IN IP4 192.168.10.141. t=0 0. m=audio 24608 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20
Re: [asterisk-users] T.38 passthru on 1.8.5
Hi, Am Dienstag, den 30.08.2011, 09:44 -0400 schrieb Fabian Borot: Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway -- asterisk -- Dialogic IMG 1010 the call starts as a voice call, the remote fax picks up and we hear the fax tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is passed back to the user side [quintum gateway] whcih reply with 200 OK with t.38 and the nothing else happens. After 20 secs of inactivity the quintum sends another Invite with voice only and then a BYE. We do see that the quintum sends a lot of messages like this from the quintum's IP [192.168.1.18] but we do not see that asterisk sends the packages to the destination UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) we have this settings on sip.conf faxdetect = yes t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] When we send the fax from the quintum to the Dialogic IMG the fax works 100% of the times. I enabled fax set debug on and udptl set debug on but the console does not show almost anything but the udptl packets shown above. What else should I do?Any ideas/help is greatly appreciated I assume there is some NAT/firewall on the way? (The 192.30.189.146 is public internet and 192.168.1.18 is RFC1918). Be sure to let udptl pass through your firewall/NAT. Udptl is using a port range of its own (different from the range used for audio). Audio is always symmetric. Audio in one directions implies audio in the opposite direction. A NAT Gateway open for one direction is normally no problem for audio. T.38 is not symmetric. One side sends and waits for some kind of ACK. The other side waits for data. If the sending side gets through the NAT Gateway all is fine. If the sending side is not getting through, the transmission is aborted (after timeout). HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8.5 Voicemail duration incorrect
Hi Robert, Am Donnerstag, den 25.08.2011, 13:28 -0400 schrieb Robert Huddleston: https://issues.asterisk.org/jira/browse/ASTERISK-16981 Thank You for the link. I already found it a few hours later. I put some debug output in the code and I think I found the location of the issue, but I currently do not know, how to fix it. (See comments in jira). Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8.5 Voicemail duration incorrect
Hi, Am Mittwoch, den 24.08.2011, 13:18 -0400 schrieb Robert Huddleston: Anyone else seen this? I saw a jira but was in feedback status.. I just checked with a voicemail of 60 seconds. It was reported in .txt-file with a duration of 19 seconds. So there is a bug. Do You have a link to the Jira issue? Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with *8 Pickup
Hi, Am Montag, den 15.08.2011, 10:18 +1200 schrieb Alec Davis: If you time the *8 just right so it is being handled during the end of the Dial then I got: [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL Does this happen when using the Pickup() application as well, or is it specific to *8? Another way to get the same result, leaving orpaned channels, and the same message as above. Get the dialplan to execute Dial(SIP/phone1SIP/phone2) Then simultaneously dial *8 from 2 other phones in the same call/pickup group as phone1 and phone2. is this bug already reported at the issue tracker/jira? Is someone working on it? Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail issue
Hi, it seems to be fixed in 1.8.4. At least I can't reproduce it there. Karsten Am Mittwoch, den 15.06.2011, 09:29 -0400 schrieb Mike: The same issue was present in 1.6 a few weeks ago and is fixed in latest 1.6. Maybe latest 1.8.4 does not have this issue. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell Sent: Wednesday, June 15, 2011 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; brya...@zktech.com Subject: Re: [asterisk-users] Voicemail issue I'm on 1.8.3.3 and it does the same thing. Once you log back in it says you have a message. You press 1 to play and she just says First then gives you options to delete, save etc. The message is in the INBOX as msg0001.wav currently. __ From: Alec Davis Sent: Wed 6/15/2011 4:12 AM To: brya...@zktech.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Voicemail issue https://issues.asterisk.org/jira/browse/18998 may have the answer, particularly the patch bug18998-1.8.2.3.diff.txt Alec __ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Wednesday, 15 June 2011 12:11 p.m. To: brya...@zktech.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail issue Ok here is a step by stop on how I can repeate the stuck voicemail box bug. Now how do I fix it? Again I am on version 1.8.2.3 build. Can some one with a newer build test and tell me if they get the same results? Example: All testing has been done with a single message in the inbox. User has a message in their inbox They call in they listen to the message. They press 9 to save the message. They select to save the message back to the 0 folder (inbox) The system changes the messages index from to index 0001 The user hangs up The system leaves the message as index 0001 The user calls in again and it says they have messages but because there is no index so they cant get at any messages in that folder. This explains over 50 instances where voicemails would get stuck in boxes with no indexed message. How do I fix this issue ASAP? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 __ From: Bryant Zimmerman brya...@zktech.com Sent: Tuesday, June 14, 2011 5:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail issue Hey all I am having instances where voicemail boxes will have a 1 message and no 0 message this causes the user to be told that they have a message that they can't get at. If I renumber the messages manually to start with the 0 numbering then the user can get their messages. What could be causing this and how can I get it out of the system. Is there a patch I can apply to the system or is there a know fix for this issue. Right now I am stuck on this version because of some bugs in the current release that are show stoppers. I am on 1.8.2.3 build. Thanks Bryant Zimmerman (ZK Tech Inc.) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] how to know length of file in seconds
Hi, Am Dienstag, den 07.06.2011, 17:07 -0400 schrieb Paul Belanger: On 11-06-07 02:31 AM, virendra bhati wrote: Hi List, Is there any way by which we can get the length of any recorded files into seconds ? $ sox foo.wav -e stat just a remark for people using newer(?)/other version of sox: In version v14.3.0 (ubunto lyquid lynx) or v14.3.1 (Debian Squeeze) the above command results in an error. You can use sox foo.wav --null stat instead. Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel
Hi randall, Am Mittwoch, den 01.06.2011, 10:00 +0200 schrieb randall: i get the following errors: pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 Your telco provider has crc on or off , that is not matching with your server cross check with them. and this problem solve 4 problems thanks for the reply, what is crc (same as crc-4?) and where can i set this? same crc or crc4 -- adding crc as follows, span=1,1,0,ccs,ami,crc, causes DAHDI to not load at all As I can see from Your first post, You are using BRI in point-to-multipoint mode. On BRI lines there is nothing like CRC/CRC4 and that is the reason, why the config is not loading any more. On a PTMP line there may be some CRC-errors from time to time, when the provider shuts down the line, which is normal in some countries. But this has nothing to do with Your initial problem. Unfortunately I don't know a solution for Your problem. It may be a hardware issue. HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] function Echo() doesn't work
Hi Felix, Am Mittwoch, den 16.02.2011, 12:47 +0100 schrieb Felix Dong: Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. are You trying to echo between local phones or is it a external call via some VoIP Provider? In latter case: do You forward the RTP traffic from external to Your asterisk? The relevant ports can be configured in rtp.conf. Configure at least 4 ports per connection. Configure port forwarding for this range of UDP ports in Youe NAT-device (e.g. router or firewall). HTH, Karsten best regards, Felix -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with 1.8 and BlindTransfer
Hi, Am Donnerstag, den 02.12.2010, 11:02 -0500 schrieb Bryant Zimmerman: Replys from Bryant On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com wrote: I am having issues with Blind Transfer on asterisk 1.8 What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS? Verison 1.8.0, Suse 11.1 There was an issue with blind transfer in 1.8.0, fixed in SVN (and maybe in 1.8.1 ?) See https://issues.asterisk.org/view.php?id=18185 HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.0-rc5: Blind transfer failed, SIP REFER Method
Am Donnerstag, den 21.10.2010, 16:27 +0200 schrieb Karsten Wemheuer: Hi, I setup an asterisk system (version 1.8.0-rc5). While using a SIP only environment I discovered a problem using blind transfer. The phones are SNOM or Aastra and are using the SIP REFER Method. The following is working: User A calls user B, B accepts the call, user A than transfers to user C The following is NOT working: User A calls user B, B accepts the call, user B than transfers to user C The call is terminated by asterisk without any warnings or error message in the CLI. Looking at Events on AMI, I can see in the first case an Event Newchannel with a Channel: AsyncGoto... followed by an Event Masquerade in prior to the Transfer. These events are missing in the second case. Is this a new bug or do I something wrong? Shall I open an issue on the tracker? for the archives: This behavior is still the same in 1.8.0. There is a workaround available (issue #18185) Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.0-rc5: Blind transfer failed, SIP REFER Method
Hi, I setup an asterisk system (version 1.8.0-rc5). While using a SIP only environment I discovered a problem using blind transfer. The phones are SNOM or Aastra and are using the SIP REFER Method. The following is working: User A calls user B, B accepts the call, user A than transfers to user C The following is NOT working: User A calls user B, B accepts the call, user B than transfers to user C The call is terminated by asterisk without any warnings or error message in the CLI. Looking at Events on AMI, I can see in the first case an Event Newchannel with a Channel: AsyncGoto... followed by an Event Masquerade in prior to the Transfer. These events are missing in the second case. Is this a new bug or do I something wrong? Shall I open an issue on the tracker? Thanks for any hints, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_genconf
Hi, Am Mittwoch, den 20.10.2010, 01:54 -0200 schrieb Flavio Miranda: Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Just one more question, what it means the RED under alarms when I type dahdi show status. It should be OK? the RED-alarm usually means line disconnected or something similar. You should check your wiring or contact Your provider. If You want to use the line, the status should be listed as OK. HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Kernel panic (asterisk 1.8.0-rc3, dahdi-linux-2.4)
Am Samstag, den 16.10.2010, 14:00 -0500 schrieb Shaun Ruffell: On 10/16/10 12:47 PM, Karsten Wemheuer wrote: Hi, Am Freitag, den 15.10.2010, 14:34 -0500 schrieb Shaun Ruffell: On 10/15/2010 04:00 AM, Karsten Wemheuer wrote: I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN card. The system starts without any errors. I discovered a severe issue. The kernel panics on a very small load. The first call normally gets through. If I start the second or third call and sometimes when I terminate the first call, the system panics (Oops text on console). After solving some difficulties (the relevant part of the Oops text scrolls out of the monitor, no serial interface), I get the text via netconsole. It seems to me, that the panic occurred in oslec (function oslec_update). But maybe I am wrong with this. In the oslec code there is a patch to enable MMX. After switching this off, the problem disappeared. AFAIK the cpu supports mmx. Where should I address this issue to? Is it a known issue? Do you have CONFIG_DAHDI_MMX defined in include/dahdi/dahdi_config.h? No, I don't think so. (This is from memory, I currently have no access to the test system). But there was a patch (I think from debian packages, original from bug tracker (dahdi_mmx_auto.diff from http://bugs.digium.com/view.php?id=13500)) which enables mmx at least for the echo canceler oslec (I think). Disabling this patch let the kernel panic disappear. Hmmm...I can't be certain since there are many parts coming from out of the tree here (besides just olsec itself), but looking at https://issues.asterisk.org/file_download.php?file_id=22366type=bug doesn't *seem* right. It appears that DAHDI_USE_MMX is exported and therefore the olsec in git://gitorious.org/dahdi-extra/dahdi-extra.git uses the MMX instructions, but -DCONFIG_DAHDI_MMX is only added onto CFLAGS_zaptel_base.o and not CFLAGS_dahdi_base.o. Therefore, oslec most likely is killing the FPU registers since it believes that dahdi-base.c is taking care of saving and restoring them by hand. I would recommend changing CFLAGS_zaptel_base.o to CFLAGS_dahdi_base.o, or hand edit include/dahdi/dahdi_config.h to make sure CONFIG_DAHDI_MMX is defined and see if you still get the crash. I changed CFLAGS_zaptel_base.o to CFLAGS_dahdi_base.o as You recommended and it seems to work now. No crash anymore. Thanks, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Kernel panic (asterisk 1.8.0-rc3, dahdi-linux-2.4)
Hi, Am Freitag, den 15.10.2010, 14:34 -0500 schrieb Shaun Ruffell: On 10/15/2010 04:00 AM, Karsten Wemheuer wrote: I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN card. The system starts without any errors. I discovered a severe issue. The kernel panics on a very small load. The first call normally gets through. If I start the second or third call and sometimes when I terminate the first call, the system panics (Oops text on console). After solving some difficulties (the relevant part of the Oops text scrolls out of the monitor, no serial interface), I get the text via netconsole. It seems to me, that the panic occurred in oslec (function oslec_update). But maybe I am wrong with this. In the oslec code there is a patch to enable MMX. After switching this off, the problem disappeared. AFAIK the cpu supports mmx. Where should I address this issue to? Is it a known issue? Do you have CONFIG_DAHDI_MMX defined in include/dahdi/dahdi_config.h? No, I don't think so. (This is from memory, I currently have no access to the test system). But there was a patch (I think from debian packages, original from bug tracker (dahdi_mmx_auto.diff from http://bugs.digium.com/view.php?id=13500)) which enables mmx at least for the echo canceler oslec (I think). Disabling this patch let the kernel panic disappear. Maybe Alex suggestion points in an interesting direction. The kernel is indeed compiled with CONFIG_PREEMPT=y. If I have enough time, I'll try compiling the kernel with another preemption model (voluntary or no) and patch applied. Thanks, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Kernel panic (asterisk 1.8.0-rc3, dahdi-linux-2.4)
Hi, I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN card. The system starts without any errors. I discovered a severe issue. The kernel panics on a very small load. The first call normally gets through. If I start the second or third call and sometimes when I terminate the first call, the system panics (Oops text on console). After solving some difficulties (the relevant part of the Oops text scrolls out of the monitor, no serial interface), I get the text via netconsole. It seems to me, that the panic occurred in oslec (function oslec_update). But maybe I am wrong with this. In the oslec code there is a patch to enable MMX. After switching this off, the problem disappeared. AFAIK the cpu supports mmx. Where should I address this issue to? Is it a known issue? Here comes one example for the oops: /- BUG: unable to handle kernel NULL pointer dereference at (null) IP: [c0103dd6] __math_state_restore+0x56/0x90 *pde = Oops: [#1] PREEMPT SMP last sysfs file: /sys/module/configfs/initstate Modules linked in: netconsole configfs dahdi_echocan_oslec echo capifs loop wcb4xxp rtc_cmos i2c_i801 rtc_core dahdi 8250_pnp 8139too floppy 8250 rtc_lib mii serial_core i2c_core processor pcspkr rng_core button ide_pci_generic ide_core sd_mod crc_t10dif thermal [last unloaded: netconsole] Pid: 1268, comm: clip.agi Not tainted 2.6.35.4 #1 P4Dual-915GL/P4Dual-915GL EIP: 0060:[c0103dd6] EFLAGS: 00010046 CPU: 0 EIP is at __math_state_restore+0x56/0x90 EAX: EBX: c5b2 ECX: cd461960 EDX: ESI: cd461960 EDI: c01045a0 EBP: 0080 ESP: c5b21cb0 DS: 007b ES: 007b FS: 00d8 GS: 00e0 SS: 0068 Process clip.agi (pid: 1268, ti=c5b2 task=cd461960 task.ti=c5b2) Stack: c5b21cd0 0027 c01045a0 c01045e5 0200 cfadd500 c0432273 0 cfadd500 cfadd200 0008 0027 0080 0080 cf33fa00 007b 0 007b c02d00d8 00e0 d0ae2153 0060 00010002 005a Call Trace: [c01045a0] ? do_device_not_available+0x0/0x60 [c01045e5] ? do_device_not_available+0x45/0x60 [c0432273] ? error_code+0x73/0x80 [c02d00d8] ? DAC960_V1_ProcessCompletedCommand+0x1108/0x1510 [d0ae2153] ? oslec_update+0xe3/0x5c0 [echo] [d0aeb038] ? echo_can_process+0x28/0x40 [dahdi_echocan_oslec] [d0aeb010] ? echo_can_process+0x0/0x40 [dahdi_echocan_oslec] [d0a08a18] ? dahdi_ec_span+0x268/0x2a0 [dahdi] [d0a9136c] ? b4xxp_interrupt+0x11c/0x358 [wcb4xxp] [c0175ded] ? handle_IRQ_event+0x2d/0xc0 [c02dd71d] ? scsi_decide_disposition+0x16d/0x180 [c0177b85] ? handle_fasteoi_irq+0x65/0xd0 [c0105a55] ? handle_irq+0x15/0x30 [c01050a7] ? do_IRQ+0x47/0xc0 [c0103d30] ? common_interrupt+0x30/0x40 [c01300e0] ? load_balance+0x550/0x7d0 [c0431614] ? _raw_spin_unlock_irq+0x4/0x20 [c012d9ba] ? finish_task_switch+0x3a/0x90 [c042f5c9] ? schedule+0x1c9/0x520 [c0103d30] ? common_interrupt+0x30/0x40 [c042facf] ? preempt_schedule+0x2f/0x50 [c0198a60] ? do_wp_page+0x160/0x960 [c0199c02] ? handle_mm_fault+0x5d2/0xaa0 [c01244b0] ? do_page_fault+0x0/0x370 [c01245f0] ? do_page_fault+0x140/0x370 [c01b7b2f] ? copy_strings+0x17f/0x1a0 [c01b935e] ? do_execve+0x2be/0x310 [c01b935e] ? do_execve+0x2be/0x310 [c010aa80] ? sys_execve+0x40/0x70 [c01244b0] ? do_page_fault+0x0/0x370 [c0432273] ? error_code+0x73/0x80 Code: 89 c2 0f ae 2f 85 c9 75 27 83 4b 0c 01 80 86 98 00 00 00 01 8b 1c 24 8b 74 24 04 8b 7c 24 08 83 c4 0c c3 66 90 8b 86 50 02 00 00 0f ae 08 eb d9 e8 c0 ed 01 00 90 83 c8 08 e8 c7 ed 01 00 90 b8 EIP: [c0103dd6] __math_state_restore+0x56/0x90 SS:ESP 0068:c5b21cb0 CR2: ---[ end trace 65c27cd3a6b7bd8a ]--- \- Thanks, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8: Warning messages in CLI while putting a SIP-Call on hold
Am Mittwoch, den 06.10.2010, 15:11 +0200 schrieb Karsten Wemheuer: Hi, while testing current release candidate 1.8.0-rc2 I stumbled on a weird behavior. I did not find any hints in the archives or at the bug tracker. Two SIP-Clients are connected (both on the local net, no NAT). The RTP stream flows directly between the phones. If I set phone A on hold, the music on hold is played. On the CLI I see the following message running: WARNING[2470]: res_rtp_asterisk.c:1939 bridge_p2p_rtp_write: RTP Transmission error of packet to (null): Invalid argument The message is running until the phones are connected again. In the meantime the CLI is nearly unusable. This does not happen, if I configure asterisk to stay in the media path. for the archives: This behavior seems to be fixed in 1.8.0-rc3. Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8: Warning messages in CLI while putting a SIP-Call on hold
Hi, while testing current release candidate 1.8.0-rc2 I stumbled on a weird behavior. I did not find any hints in the archives or at the bug tracker. Two SIP-Clients are connected (both on the local net, no NAT). The RTP stream flows directly between the phones. If I set phone A on hold, the music on hold is played. On the CLI I see the following message running: WARNING[2470]: res_rtp_asterisk.c:1939 bridge_p2p_rtp_write: RTP Transmission error of packet to (null): Invalid argument The message is running until the phones are connected again. In the meantime the CLI is nearly unusable. This does not happen, if I configure asterisk to stay in the media path. Is this a new bug or do I something wrong? File sip.conf looks like this: [general] bindaddr = 0.0.0.0 disallow = all allow = alaw allow = ulaw language = de allowguest = no fromdomain = 192.168.10.70 tos_sip = 96 tos_audio = 184 [katrin] type = friend host = dynamic callerid = Katrin Wemheuer 200 context = Standard mailbox = 200 [max] type = friend host = dynamic callerid = Max Müller 245 context = Standard mailbox = 245 Thanks, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HDLC Bad FCS (8) on Primary D-channel
Hi, Am Freitag, den 11.06.2010, 11:54 +0100 schrieb Gareth Blades: Olivier wrote: Hello, I've got a running system in which logs are full of messages such as: [Jun 10 07:24:14] NOTICE[2414] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 The strange thing is those messages are coming from a single span. My setup is : Asterisk 1.6.1.18 Junghanns OctoBRI with wcb4xxp driver libpri 1.4.10.2 dahdi 2.3.0 3 BRI lines in PtMP mode What does this PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 roughly mean ? Why could it happen on a single port and not on the others ? Regards Basically it means that one of the messages it received on the PRI D channel failed the checksum. I take it that in your span command you have 'crc4' or similar specified as an option for all of your spans? If thats the case its probably a faulty port on the card, cable, or a card in the local telephone exchange, AFAIK CRC4 is for PRI only. The setup of Olivier is BRI in PTmP mode. Many providers drive Layer 1 down in case of inactivity. Maybe the driver has a problem with such lines. In this case talk to your provider to disable this setting. HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HDLC Bad FCS (8) on Primary D-channel
Hi Olivier, Am Freitag, den 11.06.2010, 14:27 +0200 schrieb Olivier: 2010/6/11 Karsten Wemheuer k...@gmx.de Hi, Am Freitag, den 11.06.2010, 11:54 +0100 schrieb Gareth Blades: Olivier wrote: Hello, I've got a running system in which logs are full of messages such as: [Jun 10 07:24:14] NOTICE[2414] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 The strange thing is those messages are coming from a single span. My setup is : Asterisk 1.6.1.18 Junghanns OctoBRI with wcb4xxp driver libpri 1.4.10.2 dahdi 2.3.0 3 BRI lines in PtMP mode What does this PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 roughly mean ? Why could it happen on a single port and not on the others ? Regards Basically it means that one of the messages it received on the PRI D channel failed the checksum. I take it that in your span command you have 'crc4' or similar specified as an option for all of your spans? If thats the case its probably a faulty port on the card, cable, or a card in the local telephone exchange, AFAIK CRC4 is for PRI only. The setup of Olivier is BRI in PTmP mode. Many providers drive Layer 1 down in case of inactivity. Maybe the driver has a problem with such lines. Would that explain why only a single port is hit ? No, except if this port is configured differently from the others (on the provider site)... This port is the 2nd and the dialing pattern is DAHDI/g1 which means start with channel 1 on span 1, then channel 2 on span 1, then channel 4 on span 2, . Ok, but dialing is Layer 3. You are observing layer 2 errors on the D-Channel (LAPD-protocol). What I observed is that provider sends incoming calls alternatively to each span : if an inbound call comes through span 2 (channel 4 or 5), then the provider would send the next one to span 3 (channel 7 or 8) if available, etc ... To my experience a provider do not send incoming calls to different ports on PTmP lines. Each line gets his own numbers, there is no overflow (at least in germany). But again: Your original problem are layer 2 errors. Reasons could be: - line broken - port broken - driver do not handle layer 1 down in case of inactivity I'll try to swap cables and see if messages are moving from span 2 to another span. Good idea. Have a nice weekend. Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP RTP ports not released when channel is hung up
Hi, Am Donnerstag, den 18.02.2010, 10:49 +0100 schrieb Armin Schindler: On Tue, 16 Feb 2010, Armin Schindler wrote: On Tue, 16 Feb 2010, Marcus Hunger wrote: Hi, did you see this one: https://issues.asterisk.org/view.php?id=16774 ? It looks related to your issue. Oh thanks, I missed that one. It really looks related. I have added a note. Now I know how to reproduce the problem. I added this as note to 16774 as well: Start SIP client to register at asterisk, then disconnect that SIP phone from network. In the time the registration is still active in asterisk, call this phone. Asterisk will send INVITEs (of course with no answer), then hangup after about 30 seconds. The asterisk channels are released, but the sip channel for that Init: INVITE is not released. For now, I can confirm this with 1.4.28 only as I have not tested other versions yet. With version 1.4.29 I can't reproduce it the way You described it. If the caller hangs up before * times out the INVITE, the ressources are freed (SIP-channel and RTP-Ports). If * times out first, the ressources are freed some time later ( 1 minute). Karsten -- _ -- 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] Trouble getting feature codes to work
Hi, Am Donnerstag, den 21.01.2010, 21:08 -0500 schrieb hugolivude: Hi, I'm having trouble getting feature codes to work in Asterisk 1.4.21.2. Features.conf contians this: blindxfer=## atxfer=*2 automon=*1 disconnect=** I'm really most interested in getting disconnect to work so that I hear Goodbye when I press ** during a call connected this way in my dial plan: exten = 1,n,Dial(SIP/14168724...@6135551212-sw1|120|gtT) exten = 1,n,Playback(vm-goodbye) The call works fine and the CLI tells me that ** is an active feature: Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call * ** Park Call When I press ** during a call though, nothing appears in the CLI (verbosity = 4). I do it very quickly so I don't believe timeout is an issue. As DTMF recognition is not the problem (as You told in the other post), You can check two other things: 1) Exclude the timing issue: Are the other 2-character feature codes working? What about testing with a 1-character code setting or with a featuretimeout in the conf-file (I believe the default is very short) 2) If this is a sip-to-sip call, check if asterisk stays in the audio path (you can check it with a network sniffer like tcpdump or wireshark). HTH, have a nice weekend, Karsten -- _ -- 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] DTMF detection on dahdi with b4xxp (again, some more details)
Hi, Am Dienstag, den 05.01.2010, 15:38 +0100 schrieb Christian Theune: Hi, I tried again getting DTMF detection on my ISDN devices with dahdi going again. I used the channel debug to see whether asterisk sees the frames and detects them as DTMF. Interestingly here's what works: 1. GSM phone - chan_dahdi g1 - asterisk - can_sip - SIP phone Both the GSM phone and the SIP phone can issue DTMF that will be detected as features (transfer) 2. GSM phone - chan_dahdi g1 - asterisk - chan_dahdi g4 - ISDN phone The GSM phone can issue DTMF that will be detected. The ISDN phone won't. (That's my issue.) I don't see any messages of asterisk recognizing the DTMF frames when pressing the keys. I do hear the DMTF sound on both phones. 3. ISDN phone - chan_dahdi g4 - asterisk - chan_dahdi g1 - GSM phone The ISDN phone can issue DTMF that will be recognized and so does the GSM phone. So. When the ISDN phone is receiving a call on g4 its DTMF sounds won't be recognized. OTOH when the GSM phone on g1 is being called it's sounds are recognized. I *think* there are two possibilities to transfer DTMF on ISDN: - as audio on B-Channel - as Key-Press events (Info-Elements) on D-Channel DTMF on GSM can not be signalled as audio (because of codec with high compression). I guess in case GSM = asterisk via chan_dahdi g1 in Your example, the DTMF is signalled as Info-Elements on D-Channel. I guess in the cases where Your DTMF is not working, audio path is used. In this case DTMF detection is done by DSP-Software. Look for the relaxdtmf statement (in case of zaptel this worked for me in a simmilar scenario). HTH, Karsten ___ -- 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] Unable to set TOS to 184?
Hi Bart, Am Donnerstag, den 29.10.2009, 16:36 -0700 schrieb Bart Fisher: I don't understand this message: [2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos: Unable to set TOS to 184 You did not tell us, which version of asterisk You are running. The kernel restricts setting of high ToS-bits for non-root users. To allow a process to run as non-root and be able to set these bits, there is the possibility to use 'capabilities'. This feature was implemtented and fixed in the past (issues 7074 and 14004 at issues.asterisk.org). I found one post that says to run at boot: #!/bin/bash /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 4569 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 1:2 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 5060 -j DSCP --set-dscp-class ef In case Your iptables is working, I think You can ignore the warning. HTH, Karsten ___ -- 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] Calling non-extension numbers issue
Hi, Am Montag, den 29.06.2009, 10:35 -0400 schrieb Kayton Sapale: That's the strange thing. Nothing shows when monitoring the service in debug. On the phone, however, I do see a connection time-out error. I guess this might indicate that the device is attempting to connect to the service in a way different from when just dialing an extension? If You don't see anything on the command line of *, there might be an issue with Your phone settings. I don't know anything about the nokias, but I *think* it might be possible, that the phone connects to anything other than Your * box in case of the outbond number. AFAIK the * sends a 404-Error back on an non existing extension. In this case the phone would not show up a connection time-out. So I would check the settings on the phone. Or maybe You could do a network trace with tcpdump or ngrep to double check, that the phone really tries to connect to *. HTH, Karsten ___ -- 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 / Hylafax
Hi Michael, Am Samstag, den 13.12.2008, 23:09 +1300 schrieb Michael: For some odd reason the call registration issue doesn't seem to stop it working, except a few seconds after Hylafax answers the call it hangs up, I suspect because Asterisk only supports T38 pass through. Here is my data path PSTN = T.38 aware SIP provider = Internet = My machine (Asterisk) = IAX modem on localhost = Hylafax. This path will not work. As You mentioned, * supports T38 path through only. In Your setup there will be a conversion on the * box between T38 via SIP provider and IAX (which uses G711 codec in this case). To make it work, use newer versions of t38modem and replace the iaxmodem with it. Newer versions of t38modem supports SIP, so that Your path will be PSTN = T.38 aware SIP provider = Internet = My machine (Asterisk) = T38-modem on localhost = Hylafax. HTH, Regards Karsten ___ -- 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] One Way Audio Problem
Hi, Am Donnerstag, den 16.10.2008, 09:37 +0800 schrieb GNUbie: Hello Karsten, On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote: Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? The SNOM 300 is using the NET interface beside the DC 5V port to connect to the LAN. The Asterisk box is using the eth1 to connect to the LAN. As per your instruction, below is my /etc/asterisk/sip.conf : - - - s n i p - - - [general] realm=pbx.domain.com bindport=5060 bindaddr=0.0.0.0 rtptimeout=60 disallow=all allow=ulaw allow=alaw allow=gsm externip=pbx.domain.com localnet=192.168.101.0/255.255.255.0 jbforce=yes allowtransfers=yes maxexpiry=3600 minexpiry=1800 videosupport=no [internal-phones](!) type=friend host=dynamic context=family dtmfmode=rfc2833 insecure=port,invite canreinvite=no nat=no qualify=yes port=5060 [102](internal-phones) username=102 secret=102 callerid=GNUbie102 [EMAIL PROTECTED] - - - s n i p - - - Thanks for the information. In an earlier post You told us, that the local phones talk to asterisk on eth1 using 192.168.101.0 network. Could You please double check, that the phone did not try to register on another IP? The asterisk is IIRC on a dual homed machine. Is Your phone using a DNS lookup to find the asterisk? To which address is that lookup resolved? Another hint: Is Your SNOM using some sort of STUN to lookup an public address? (Just to eliminate some things). HTH, Karsten ___ -- 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] One Way Audio Problem
Hi, Am Montag, den 13.10.2008, 10:00 +0800 schrieb GNUbie: Hello Gordon, On Mon, Oct 13, 2008 at 2:22 AM, Gordon Henderson [EMAIL PROTECTED] wrote: You mention the SIP phone being inside the LAN. Where is the Asterisk box? It is the main gateway of the IP phones and my laptop to the Internet. In this case, the eth1 of the Asterisk box is connected to the LAN and eth0 is connected to the Internet. IME: One-way audio problems are almost always casued by NAT gateways and/or incorrect NAT settings in sip.conf and/or incorrect IP address or extenal proxy settings in the SIP phone. I don't think NAT is involve on this one way audio problem. Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? Regards, Karsten ___ -- 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] cli commands missing
Hi Eric, Am Sonntag, den 12.10.2008, 18:06 -0700 schrieb Eric Fort: resolve.conf and dns is working. The problem persists. /var/log/asterisk/messages shows a few notices and warnings on res_smdi.c, res_musiconhold.c, and usbradio.c. when I disable loading of these in modules.conf asterisk crashes on load. Eric try to start asterisk in foreground. First stop it and then start it on the shell: asterisk -vv You will see all the messages while autoloading. Some of them might tell You, why a module is not loaded. HTH, Regards, Karsten ___ -- 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] Read one or X DTMF
Hi Ruddy, Am Freitag, den 12.09.2008, 21:38 -0400 schrieb Ruddy Gbaguidi: But user just needs to enter * instead of *# We are doing this because 80% of the callers already have an account, so, instead of playing : If you have an account, press 1, if not press 2 we prefer to play Enter you account now or press * if you don't have any the 'read' application I mentioned needs a '#' to be pressed. Otherwise You have to wait for a timeout. I think You'd better take the solution Tony mentioned. Regards, Karsten ___ -- 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] Read one or X DTMF
Hi, Am Freitag, den 12.09.2008, 11:03 -0400 schrieb Ruddy Gbaguidi: Hi all I'm just having a problem now and I don't have any idea how to do this. It is pretty simple. When a customer calls, to speed up the navigation in the dialplan, I want something like Welcome. Please enter your 10 digit customer number or press * to register So, I want to read up to 10 digits, and if the user press *, I want to go to the next extension. Do you have an idea ?? You can use the read application to get some digits. This application returns the number a user entered in a variable. If the user enters '*' the variable is set to an empty string. You can than proceed in Your dialplan. To distinguish the answers, You can use the function len. The read application is able to play a audio file. (see the doc with 'core show application read') One little hint: If You start a new thread, create a new message instead of using an old one. Your question is now part of the thread about application jack and its runtime, what is probably not what You want. Maybe some people ignore Your mail, because they are not interessted in jack... Regards, Karsten ___ -- 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] Extension not found
Hi Michel,
Am Freitag, den 12.09.2008, 17:41 +0300 schrieb michel freiha:
Dear All,
I have the following scenario...When a customer dial 111 number a beep
message will iplay in order to record and playback his voice...Else
he'll be routed to another call flow as you can see in the context
below:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? custom-recordme,111,1)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
But i have the following error when trying to dial 111:
[Sep 12 14:16:32] WARNING[30978]: pbx.c:2483 __ast_pbx_run: Channel
'SIP/michofr-093833e0' sent into invalid extension '111' in context '
custom-recordme', but no invalid handler
The above dialplan sends the call into context custom-recordme with
extension 111 and to priority 1, if the caller dials 111. For further
help we would need that context too.
Regards,
Karsten
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Re: [asterisk-users] Read one or X DTMF
Hi Ruddy, Am Freitag, den 12.09.2008, 13:22 -0400 schrieb Ruddy Gbaguidi: Thanks for the hint. Sorry about that. If I use your soution, I cannot make any difference between a user pressing * and a user that reach the timeout because he didn't enter any digit. In both cases, I will have an empty string You can use the variable EPOCH to get a timestamp before and after execution of the read application. If the difference of the two values evaluates to the timeout, the user enters nothing. Otherwise the user enters '*#' or directly the #-key without anything more. I don't know how to distinguish this two cases. Regards, Karsten ___ -- 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] problem with iaxmodem!
Hi, Am Mittwoch, den 06.08.2008, 17:24 +0200 schrieb Nadjia Boumédiène: My iax.conf looks like this: [iaxmodem] type=friend host=127.0.0.1 secret=x context=fax-out permit=127.0.0.1 disallow=all allow=ulaw after editing inittab I reload it by running: /sbin/init q I also reboot the system with shutdown -r now and I had the following message: init: Id mo respawning too fast: disabled for 5 minutes. this message indicates, that the service (identified through id mo) died short after it's start. So the init-process starts it again and again. Cause this happens to fast, it disables the restarting of the process. Having a look at Your inittab in Your first post, I would suggest to remove the -D switch from the line with faxgetty. This command line switch instructs the faxgetty to detach from terminal. In this case the init process looses contact to the process and tries to restart it. HTH, Karsten ___ -- 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] Dialplan Action on Authentication
Hi David, Am Sonntag, den 20.07.2008, 11:57 +0200 schrieb David Ashwood: Morning guys and gals, I’d like to be able to run some code when a device (soft/hardphone) authenticates to Asterisk. I remember reading somewhere that there’s the possibility of part of a dialplan can be run when a device authenticates. Does anybody have a pointer to some documentation or some pointers about the context that can be used when a device authenticates/unauthenticates to Asterisk? I’m looking for some actions to be performed on Client Authentication without using a manual authentication (using VMAuthenticate or AgentLogin). As Alex said, it is impossible to do this from dialplan. But maybe it is possible for You to use the manager API. On the manager interface there is an event fired, whenever a peer (SIP or IAX) registers. So it should be possible to logon to the manager interface, wait for the event and do some action. If You want go back to the daiplan, you can originate a call to a local channel when the event occurs. HTH, Karsten ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk seg fault
Hi, Am Mittwoch, den 25.06.2008, 08:42 -0400 schrieb Jerry Geis: I am running asterisk from svn check out from yesterday Jun 24. I started with 1.4.20, then 1.4.21 then svn. I am getting: pcm_local.h:389 snd_pcm_channel_area_addr assertion bitsofs %8 = 0 failed segment fault. I am running debian i386, on a 486 sx machine. I am connecting to the Console/DSP and then I get the seg fault. Only thing in asterisk I changed from the default was turning off codec_lpc10. Which I am not using anyway. What should I do with this error? If You are realy using a 486sx, please remember, that this CPU does not have a math copro. Maybe that's the cause for the failure. Do You have the Math-Emulation in kernel options activated? Other reasons can be some optimisations this processor doesn't support. As Tilghman wrote, the error does not occur in asterisk. According to Your second posting, I would suspect the alsa-stuff. HTH, Karsten ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with SIP, attended transfer and GROUP_COUNT
Hi,
maybe someone can give me a hint to solve the following issue. I want to
limit the calls to a specific SIP-destination. Disabling callwaiting at
the phones is not an option, because it should be configured via the *
database.
My solution uses GROUP_COUNT, which works fine most of the time. In case
of attended transfer (on SIP-basis, not via the #-mechanism of asterisk)
I have problems.
To simplfy the scenario I stripped down the dialplan to the following.
From somewhere on the wiki I am using the following context:
exten = 200,1,Set(GROUP()=${CALLERID(num)})
exten = 200,n,GotoIf($[${GROUP_COUNT(${EXTEN})} = 1]?BLOCK)
exten = 200,n,Set(OUTBOUND_GROUP=${EXTEN})
exten = 200,n,Dial(SIP/katrin)
exten = 200,n(BLOCK),Busy
This block is used for other extensions 100 and 150 respectivily. It
works fine until I am using attended transfer.
Example: kwe (Extension 100) is calling katrin (Extension 200). katrin
sets the call on hold and talks to hans (Extension 150).
At the cli I get the following result:
pbxtest*CLI group show channels
ChannelGroup Category
SIP/kwe-081bf188 100 (default)
SIP/katrin-081b70a8200 (default)
SIP/katrin-081bb020200 (default)
SIP/hans-0816b8b8 150 (default)
which seems correct to me.
In case of a transfer of kwe to hans (katrin leaving), the result is:
pbxtest*CLI group show channels
ChannelGroup Category
SIP/kwe-081bf188 100 (default)
SIP/kwe-081bf188 200 (default)
SIP/hans-0816b8b8 150 (default)
I am confused about the second line, which leads to trouble. The above
context would think, that katrin is busy. In case of a blind transfer
everything is ok (the second line does not exist)
I have tested the above with * 1.4.14, 1.4.18-rc4 and 1.4.19
Is this a bug or a feature? Am I doing something wrong or should I file
a bug report?
Thanks in advance,
Regards
Karsten
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Re: [asterisk-users] Ring back when free?
Hi, Am Freitag, den 04.04.2008, 13:03 + schrieb Tony Mountifield: In article [EMAIL PROTECTED], Faraz R. Khan [EMAIL PROTECTED] wrote: Thinking out loud: write a asterisk call file (when the calling user presses 5) which keeps on trying to connect the two. I thought about that, but the trouble is, it's not event-driven. It just keeps on trying until it runs out of retries. We realized someting like that with a call file. If a caller presses 5 store this as an open callback in a database. Place a script in the h-extension and call it with the DeadAGI-Application. This script looks up any pending callbacks for both parties of the closed connection and generates the call files. Regards, Karsten ___ -- 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] Two phones fail to agree on codec, asterisk at fault?
Hi Martin, Am Freitag, den 28.03.2008, 14:27 +0100 schrieb martin f krafft: [...] So calls are going via an asterisk bridge and the symptoms of my problem are: 1 if C450IP calls softphone, they can talk fine 2 if softphone calls C450IP, voice only goes from C450IP to softphone, not the other way around. I traced this down to the session description protocol, where there is funky stuff going on with the supported codecs each peer announces. Remember, asterisk is between them, and I set disallow=all,allow=ulaw,allow=alaw in [global]. If this isn't a typo, use [general] instead of [global]. HTH, Karsten ___ -- 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] GROUP_COUNT and Attended transfer
Hi Paul, Am Dienstag, den 05.02.2008, 10:10 +1100 schrieb Paul Hales: With some of the phones (snom, for example) you can turn off mwi, so the phone will only accept one call at a time. Much easier. PaulH Thanks for Your answer. Unfortunaly turning call waiting off is not an option for me. Some clients aren't able to switch it off and some users want to use the web gui to set the group count via the * database. Do You know, if it is a bug or a feature? Regards Karsten ___ -- 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] Mistake in the wiki's description of cmd Pickup() ?
Hi Stefan, Am Dienstag, den 05.02.2008, 10:30 +0100 schrieb Stefan Guenther: Hi, according to the description of Pickup() on page http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup I can use this command to pickup a call at a certain extensions. When I try this with e.g. exten = *8200,1,Pickup(200) I see, that You are using bristuffed *. As bristuff has its own pickup mechanism, be careful using the right one. AFAIK You have to use DPickup if You want to pickup a call by extension. In the bristuffed version of * Pickup is used with a group and DPickup is used with an extension (AFAIK). HTH, Karsten ___ -- 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] GROUP_COUNT and Attended transfer
Hi,
I want to use GROUP_COUNT to limit calls to a specific destination. From
somewhere on the wiki I am using the following context:
exten = 200,1,Set(GROUP()=${CALLERID(num)})
exten = 200,n,GotoIf($[${GROUP_COUNT(${EXTEN})} = 1]?BLOCK)
exten = 200,n,Set(OUTBOUND_GROUP=${EXTEN})
exten = 200,n,Dial(SIP/katrin)
exten = 200,n(BLOCK),Busy
This block is used for other extensions 100 and 150 respectivily. It
works fine until I am using attended transfer.
Example: kwe (Extension 100) is calling katrin (Extension 200). katrin
sets the call on hold and talks to hans (Extension 150).
At the cli I get the following result:
pbxtest*CLI group show channels
ChannelGroup Category
SIP/kwe-081bf188 100 (default)
SIP/katrin-081b70a8200 (default)
SIP/katrin-081bb020200 (default)
SIP/hans-0816b8b8 150 (default)
which seems correct to me.
In case of a transfer of kwe to hans (katrin leaving), the result is:
pbxtest*CLI group show channels
ChannelGroup Category
SIP/kwe-081bf188 100 (default)
SIP/kwe-081bf188 200 (default)
SIP/hans-0816b8b8 150 (default)
I am confused about the second line, which leads to trouble. The above
context would think, that katrin is busy.
I have tested the above with * 1.4.14 and 1.4.18-rc4
Is this a bug or a feature? Am I doing something wrong or should I file
a bug report?
Thanks in advance,
Regards
Karsten
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Re: [asterisk-users] Cisco power injector with GXP2000 phones
Hi, Am Donnerstag, den 06.12.2007, 11:30 -0500 schrieb Jon Pounder: Quoting Ricardo Carvalho [EMAIL PROTECTED]: I only see one explanation to my problem... GXP2000 phones only implement PoE mode A of the IEEE 802.3af protocol, and the power injector does only PoE mode B of the IEEE 802.3af protocol. The switch does mode A. The problem is that I can't prove this! can't find documentation with this kind of detail. If someone does, please tell. what is the difference between the modes - is it just the pins used ? Mode A means inline power. The 48 V is transferred over the wire-pairs 1/2 and 3/6 which are also used for data transfer. Before the power is switched on, endpoint and power supply must negotiate this process. Mode B means midspan power. The 48 V is transferred over the wire-pairs 4/5 and 7/8, which are unused and 10/100 networks. Mode B is not possible on GigE (afaik). Regards, Karsten ___ --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] get egress SIP call Id
Hi Ray, Am Dienstag, den 09.10.2007, 10:10 -0500 schrieb Ray Chen: Hi Philipp, Thank you for your response to my question. I am working on a project which uses Asterisk as the voice engine. I need to get the ingress and egress sip call id for a call to write call CDR. (Asterisk CDR does not meet our customer requirments). If there is no any easy way to get it I might need to create a seperate process/thread to query manager interface as you mentioned. Thanks you, Maybe You can do the trick with a Dead-AGI script. Run that script in the 'h' priority and set the Userdata field of the CDR. Than configure Your CDR to include the userdata field in the output (depends on Your CDR backend). I have not tested this, but it might be easier than hacking a new process... For the details look at the description of the * CLI: (core) show function CDR and the description of AGI: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AGI HTH, Karsten ___ --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] Configuration files inside SQLite3
Hi Mark, Am Mittwoch, den 03.10.2007, 11:15 -0500 schrieb Mark Michelson: GNUbie wrote: Hello all, Is it possible to store, read and write configuration files in an SQLite3 database instead of using the configuration files inside the /etc/asterisk/ directory? If it is then can you point me to the right documentation on how to do this or probably hints on how to do this? Thank you in advance. GNUbie It is possible to store configuration files in any relational database which has ODBC compatibility. Thus, sqlite qualifies. If you are using trunk, you won't even need to use ODBC, because Asterisk has native support for sqlite. Are You shure the native support of asterisk is for SQLite3 as the original poster asks for? AFAIK * supports SQlite (Version 2, not 3), which has a completely different API. Karsten Wemheuer ___ --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] Musiconhold instead ringing
Hi,
Am Samstag, den 08.09.2007, 09:44 + schrieb wassim darwish:
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com Date: Fri, 7 Sep 2007 19:10:04
-0500 Subject: Re: [asterisk-users] Musiconhold instead ringing On
Friday 07 September 2007 07:02:01 pm wassim darwish wrote: Hi:
When i get an incoming call, i want asterisk to make the caller hear
musicmusiconhold instead of ringing,Can any body help me with
this? my guess is you'd have to Answer() the call first, then play
moh while Dial()'ing the exten. -- Anthony - http://messinet.com -
http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967
92DC 35DC B001 4A4E
Hi:
Can you represent it in extensions Dialplan ?
Untested, but something like this should work
exten = _X.,1,Answer
extem = _X.,2,Dial(SIP/${EXTEN}|50|m(musiconhold-class)
You have to fix this to match Your existing dialplan (Extensions,
SIP-Accounts...).
It won't work without the answer-statement.
HTH,
Karsten
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Re: [asterisk-users] Slow list
Hi, Am Donnerstag, den 05.07.2007, 14:58 -0400 schrieb Andrew Kohlsmith: On Thursday 05 July 2007 2:38 pm, Doug Lytle wrote: Already did that. I use ASSP for filtering. Digium and associated mailing lists are white listed. There was only 1 attempt for deliver and there were no delays. I subscribe to 10 mailing lists (Including the dev list) and they are not having issues. By the way, the only reason I'm able to respond to your messages and I'm watching the archives at lists.digium.com I am having no issues with Digium's lists. They get a little laggy at times, but generally are fast enough. I don't agree. Your mail was delivered on 12 July (7 days delay): Received: from lists.digium.com (EHLO lists.digium.com) [216.207.245.17] by mx0.gmx.net (mx083) with SMTP; 12 Jul 2007 18:52:37 +0200 Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I6Wa3-0006CM-7P; Thu, 05 Jul 2007 14:01:31 -0500 I think, something is wrong. Currently the most recent mail I got from the list is from 11 July! Regards Karsten ___ --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] Log CODECS in CDR's
Hi Morgan,
Am Freitag, den 11.05.2007, 10:32 +0100 schrieb Morgan Gilroy:
Thanks for the pointers, I know about the Set(CDR..) function but I need
the codec that was negotiated in the Dial (once I have that its easy to
stick it into the cdrs as you pointed out).
Ie a call comes in as G729 Dial then negotiates GSM for the outbound
leg,
I want to log both these codecs in a CDR.
At the moment to find the codecs used I have to look though the sip
trace or show channels/show channel (annoying when you have 50+
channels).
Im just trying to find an easier and quicker way to keep track of the
codecs used to help with debug etc.
The closest variable iv found is, ${SIP_CODEC} Set the SIP codec for a
call
Ill see if NoOp (${SIP_CODEC}) shows the codec that was used without me
setting it though I don't think it will.
Iv looked all over and I cant find anything so it looks like I may have
to hack a ast_set_var into app_dial or chan_sip
It is untested, but maybe You can write a little AGI-Script which
accesses some channel vars. Call that AGI as a DeadAGI. A DeadAGI will
be called, if a connection terminates (connect it with the
'h'-Extension, see the wiki). I don't know if the neccessary information
is still alive at this time, but maybe it will do what You want...
HTH,
Karsten
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Re: [asterisk-users] BeroNet HFC-4S card is now detected as only 2 ports
On Fri, Apr 06, 2007 Tzafrir Cohen wrote: On Fri, Apr 06, 2007 at 01:18:24AM +0200, Henrik Woffinden wrote: Hello list, After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently detected as 2 ports instead of 4. I still load the driver as modprobe qozap ports=12 as I've always done. But now it only sees 2 ports. Output of lspci -vvv -- cut 02:01.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) Subsystem: Cologne Chip Designs GmbH Unknown device b560 Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Interrupt: pin A routed to IRQ 22 Region 0: I/O ports at ddb8 [size=8] Region 1: Memory at fcefa000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- -- cut Just a comment: the CHANGES file has the item fixed detection of miniPCI cards (qozap). Take a look at the diff between qozap/qozap.c in -d and in -e . And have a look at code parts where the hex constant 0xb5xx is mentioned. This part of the code differentiate between 2-, 4- and 8-port cards (and between the rev. 2.0 of the hardware). I found this values: CardJunghanns Beronet 2-Port 0xb556 0xb566 4-Port 0xb520 0xb560 4-Port, Rev. 2.00xb550 8-Port 0xb552 0xb562 8-Port, Rev. 2.00xb55b 0xb56b Looks like You are using 4-Port Beronet with old Hardware-Revision. I think the if clauses with the PCI-Ids fall through to some defaults which now doesn't work for Your card. The pure bristuff does not support any of the beronet cards but the old default behaviour of the code is correct for the 4 port model of beronet cards. Btw: Does someone know the PCI-ID of 4-port Beronet Rev. 2.0? Have nice easter weekend, Karsten ___ --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] chan_capi and only one B channel usable?
On 03/22/2007, Torge Szczepanek wrote: Hello list! I have a Asterisk 1.2.10 running using the package from Backports.org for Debian Sarge. I have setup chan_capi (0.6.5 from Backports) and it seems that I am only able to use on B-Channel. When trying to place the second call I get: CAPI INFO 0x34a2: No circuit / channel available capi info shows: Contr1: 2 B channels total, 1 B channels free And capi.conf is: [ISDN1] msn= isdnmode=msn incomingmsn=* controller=1 group=1 softdtmf=on accountcode= context=remote echosquelch=1 echocancel=yes echotail=64 callgroup=1 devices=2 Any ideas? If You don't have any active calls from * using ISDN, there may be other software using ISDN via the capi stack in Your box. The message capi info shows: Contr1: 2 B channels total, 1 B channels free means, that there is one B-Channel used by CAPI (not neccessarily *). HTH, Karsten ___ --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] Mini-ITX board + FXO PCI card?
Hello, Am Donnerstag, den 15.02.2007, 10:55 +0800 schrieb Leo Ann Boon: 1. The smallest mini-ITX case I found that accepts a PCI card is the Travla C138: If you used a mini-ITX with a Digium TDM400P, do you know if it fits? I didn't find its width, and apparently, the C138 will not accept a PCI card bigger than 17,52cm. The C137 can fit 2 TDM400P with the right riser. If You are using the riser card, there will be shared interrupts. The two slots of the riser card are using the same IRQ AFAIK. Karsten ___ --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] No D-channels available! Using Primary channel 16 as D-channel anyway!
Hi, maybe I am a little bit late with this answer. I take a look at Your config and the debug output. snip Provider --te11xp--- asterisk ---te11xp-- nortel merridian option 11c snip zapata.conf - context=from-pstn switchtype=dms100 signalling=pri_cpe callerid=asreceived group=1 callgroup=1 pickupgroup=1 rxgain=0.0 txgain=0.0 channel=1-15,17-31 context=from-pstn switchtype=dms100 signalling=pri_cpe callerid=asreceived group=2 callgroup=2 pickupgroup=2 rxgain=0.0 txgain=0.0 channel=32-46,48-62 able to start asterisk. Span 2 loaded beautifully, no problem or errors, but i get this WARNING[13655]: chan_zap.c:2287 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! warning message from span 1. which i have no idea what happened.. the funny part is, i am able to receive calls, no problem at all. CID and DID are pass thru (for incoming). but when i try to make a outgoing calls, i got errors PRI HANGUP CAUSE 1 from the provider. i did a pri debug span 1 for the call i dialed, below are the msg. - -- Accepting call from '124' to '42707898' on channel 0/21, span 2 -- Executing Set(Zap/52-1, CALLERID(number)=50399100) in new stack -- Executing NoOp(Zap/52-1, 50399100) in new stack -- Executing Dial(Zap/52-1, ZAP/g0/42707898||) in new stack -- Making new call for cr 32771 the dial command takes g0 as argument. But in Your zapata.conf there are configured only g1 and g2. Is this a typo? HTH, Karsten ___ --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] volume control in VoIP
Hi, On Fryday, 2007-02-02 François Delawarde wrote : Don't you think it could be an interesting feature in Asterisk? It already does transcoding, why not gain when voice flow passes through it? François. On a SIP-to-SIP-Call Asterisk is not neccessarily in the voice flow, so this does not work in any case. Karsten ___ --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] Q: Status of feature Call Deflection / Partial Rerouing in chan-capi and zaphfc
Hello, AFAIK the feature CD (call deflection) is only possible on point-to-multipoint links, is this correct? I've heard about the feature partial rerouting which should do the same on point-to-point-links. Is this implemented in either bristuff or chan-capi(-cm)? Thanks in advance, Karsten ___ --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] Problem with chan-capi: outgoing calls on two lines
Hello, while testing the following scenario, I ran into trouble: One * box with two AVM active controllers in Point-to-Point-Mode is connected to another * box with ZapHFC/Quad-BRI cards using bristuff in NT-mode. All is working fine, I can call from one box to the other and vice versa. But if I'll cut one line, it is not possible to place an outbound call from chan-capi accross the still existing line. In detail: When all lines are connected, the first two calls are placed on line 1 (which is on controller 1). The next two calls are placed on line 2 (on controller 2) If I'll cut line 2, all works as expected (I can place two calls on line 1). But if I'll cut line 1, leaving line 2 up and running, I can not place any call. The CLI tells something about Protocol error layer 1 (broken line or B-channel removed by signalling protocol) and No one is available to answer at this time (1:0/0/0) If I do the same thing in the opposite direction (Calls are initiated from the other box with bristuff in NT-mode), all works fine. What am I doing wrong (or is this a bug)? Thanks in advance, Karsten ___ --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] Problem with chan-capi: outgoing calls on two lines
Hello, On Mo, 27 Feb 2006, Armin Schindler wrote: On Mon, 27 Feb 2006, Karsten Wemheuer wrote: In detail: When all lines are connected, the first two calls are placed on line 1 (which is on controller 1). The next two calls are placed on line 2 (on controller 2) If I'll cut line 2, all works as expected (I can place two calls on line 1). But if I'll cut line 1, leaving line 2 up and running, I can not place any call. The CLI tells something about Protocol error layer 1 (broken line or B-channel removed by signalling protocol) and No one is available to answer at this time (1:0/0/0) If I do the same thing in the opposite direction (Calls are initiated from the other box with bristuff in NT-mode), all works fine. What am I doing wrong (or is this a bug)? This is not a bug, just normal behaviour. chan_capi does not know about the status of the ISDN line, it assumes to be usable when configured. So when you try to dial out chan_capi will choose a channel/line according to internal list of free channels and selects it in the CAPI request. When the driver reports an error via CAPI, chan_capi just signals this error to Asterisk. There is no logic in chan_capi to do something like: If the controller 1 isn't ready, use controller 2. Ok, I didn't know the details of the capi layer. The same happens if the b-channels are already used by another application/device. ... which would not happen on a line in point-to-point mode. And I just checked, that all works ok, if the channels are in use of asterisk itself (incomming calls). So all is ok, except that there seems to be no possibility to check the lower layer state e.g. a broken line. (In case of zap You can look at the files in /proc/zap). Thanks for Your quick response Karsten ___ --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] Problem with chan-capi: outgoing calls on two lines
Hello Armin, Am Mo, den 27.02.2006 schrieb Armin Schindler um 20:23: On Mon, 27 Feb 2006, Karsten Wemheuer wrote: Hello, On Mo, 27 Feb 2006, Armin Schindler wrote: This is not a bug, just normal behaviour. chan_capi does not know about the status of the ISDN line, it assumes to be usable when configured. So when you try to dial out chan_capi will choose a channel/line according to internal list of free channels and selects it in the CAPI request. When the driver reports an error via CAPI, chan_capi just signals this error to Asterisk. There is no logic in chan_capi to do something like: If the controller 1 isn't ready, use controller 2. Ok, I didn't know the details of the capi layer. The same happens if the b-channels are already used by another application/device. ... which would not happen on a line in point-to-point mode. No, the line mode doesn't matter. If you have another application running using CAPI, chan_capi also doesn't know about the usage. No, I have no other applications running. And on a point-to-point link there is no other device. As long as there is no outage on the line, I would not have any problems. Thanks for Your information Karsten ___ --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] MoH trouble with latest bristuff (0.3.0-PRE-1f) - SOLVED!
Hi, I answer to my own posting... On Sun, Jan 15 2006 Karsten Wemheuer wrote: Hi, I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When I activate music-on-hold on a SIP-to-SIP connection, the music sounds like in a fast-forward play mode. On the *-console I can see much lines like this: -- Silence suppression is disabled (option_silence_suppression=0 chan-timingfd=18) What's going on? With bristuff 0.3.0-PRE-1d everything works fine (but there was another issue, so I have to upgrade). The following configuration in /etc/asterisk/asterisk.conf helps: [options] silence_suppression=yes Thank's to a post of Dan Austin on this list. The messages on the console are still there, but You can disable this message by lowering the verbosity below 9. The sound is ok now. Cheers Karsten ___ --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] distorted native music on hold
Hi, On Mon, Jan 16 2006 Louis-David Mitterrand wrote: Hello, Using asterisk-1.2.1 I am trying to convert my music-on-hold files from .wav to alaw: % sox moh.wav -r 8000 -c 1 moh.al resample -ql The file sounds fine when listened with: % sox mox.al -t ossdsp /dev/dsp But when listened through asterisk with an alaw SIP phone the sound is clicky and too fast. Did I forget something in my conversion command? Are You using bristuff 0.3.0-PRE-1f? I've had the same issue. Dan Austin wrote a notice in a mail on this list, which solved the problem. Configure the following lines in /etc/asterisk/asterisk.conf: [options] silence_suppression=yes The bristuff seems to include an additional patch, which isn't stable enough. HTH, Karsten ___ --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] MoH trouble with latest bristuff (0.3.0-PRE-1f)
Hi, I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When I activate music-on-hold on a SIP-to-SIP connection, the music sounds like in a fast-forward play mode. On the *-console I can see much lines like this: -- Silence suppression is disabled (option_silence_suppression=0 chan-timingfd=18) What's going on? With bristuff 0.3.0-PRE-1d everything works fine (but there was another issue, so I have to upgrade). Thanks in advance, Karsten ___ --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] CAPI unable to handle busy()
Hello Armin, On Mo, 2 Jan 2006 Armin Schindler wrote: I don't think it is necessary to exclude it. Just build chan_capi-cm and overwrite chan_capi.so as well as remove the app_capi* modules from your installation. Armin Many thanks, it is working. Karsten ___ --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] unable to execute call file
Hello, as You are running two processes handling SIP (asterisk and openser), I think the Call-File addresses the wrong instance. If Your callfile contains a line like Channel: SIP/accountname try something like Channel: SIP/[EMAIL PROTECTED]:port where ipaddress and port addressing the responable instance. HTH, Karsten ___ --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] CAPI unable to handle busy()
Hello, first of all, I say Happy New Year to this list! While using asterisk 1.2.1 with bristuff-0.3.0-PRE-1d (which includes chan_capi 0.4.0-PRE1), I ran into the following problem. I want to signal busy to an incoming call, but that doesn't work. The dialplan looks like this: exten = 22715292,1,Busy (The extension is ok and works fine, if I use other applications like Dial) The result is: -- creating pipe for PLCI=0x101 msn = 22715292 sent ALERT_REQ PLCI = 0x101 -- Executing Busy(CAPI/contr1/22715292-13, ) in new stack -- started pbx on channel (callgroup=0)! The caller hears still ringing signal. If I replace Busy with Busy(2), the following happens: -- creating pipe for PLCI=0x101 msn = 22715292 sent ALERT_REQ PLCI = 0x101 -- Executing Busy(CAPI/contr1/22715292-14, 2) in new stack -- started pbx on channel (callgroup=0)! == Spawn extension (incoming, 22715292, 1) exited non-zero on 'CAPI/contr1/22715292-14' -- CAPI Hangingup -- removed pipe for PLCI = 0x101 But again, the calling site gets no busy-signalling. If I use hangup(17) instead of busy() (which should be the same as 17 is the value for the busy condition), I get the following result: -- creating pipe for PLCI=0x101 msn = 22715292 sent ALERT_REQ PLCI = 0x101 -- Executing Hangup(CAPI/contr1/22715292-15, 17) in new stack == Spawn extension (incoming, 22715292, 1) exited non-zero on 'CAPI/contr1/22715292-15' -- CAPI Hangingup sent CONNECT_RESP for PLCI = 0x101 -- removed pipe for PLCI = 0x101 -- started pbx on channel (callgroup=0)! Jan 2 14:00:36 ERROR[1143]: chan_capi.c:1237 pipe_frame: wrote -1 bytes instead of 48 The calling site will see a normal call clearing. Hardware is a FritzPCI! (AVM). If I do the same things with a HFC-based card and chan_zap, both version (busy() and hangup(17)) are working fine. Any helping hints are welcome! Thanks! Karsten ___ --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] CAPI unable to handle busy()
Hello Armin, On Mo, 02.01.2006 Armin Schindler wrote: On Mon, 2 Jan 2006, Karsten Wemheuer wrote: Hello, first of all, I say Happy New Year to this list! While using asterisk 1.2.1 with bristuff-0.3.0-PRE-1d (which includes chan_capi 0.4.0-PRE1), I ran into the following problem. I want to signal busy to an incoming call, but that doesn't work. The dialplan looks like this: exten = 22715292,1,Busy (The extension is ok and works fine, if I use other applications like Dial) chan_capi from junghanns/bristuff does not support that. I suggest using the new chan_capi-cm-0.6.2 thanks for the quick response. How can I implement bristuff-patch and Your new chan_capi? I need a version with both, ZAP-HFC and CAPI. So the question is, how can I exclude chan_capi from the bristuff-patches? Any Ideas? Thanks Karsten ___ --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
