Re: [asterisk-users] Subscribe Problem - Zombie Channel
Hi Brian, Did you ever figure out what's causing this, and how to deal with it? I'm seeing the same behavior with call-pickups (it's rare, but it's happened a few times) on Asterisk 1.6.1.11 Did you figure out a way to get rid of the channel without restarting? Regards, Örn On Wed, Jul 28, 2010 at 9:45 PM, dotnetdub wrote: > > > On 28 July 2010 21:42, Stefan Schmidt wrote: >> >> dotnetdub schrieb: >> > Hi List, >> >> > core show channels >> > Channel Location State Application(Data) >> > >> > SIP/102--08e1 *8@from-inside Down (None) >> > SIP/102--08d6 *8@from-inside Ring (None) >> > SIP/102--08d7 *8@from-inside Ring (None) >> > 3 active channels >> > 0 active calls >> > >> > The only way to free them up is to force a restart. >> > >> > restart now >> > >> > Any clues on how I can debug this and try to sort it or even if anyone >> > has come across this. >> > >> > Many thanks in advance. >> > >> > Brian >> > >> > >> hello, >> >> you should recompile asterisk with DEBUG CHANNEL LOCKS flag and i think >> you will see some locks when this happens with core show locks. >> how do you make the pickup? do you use an extension *8 for this, or just >> the feature for pickup in features conf? >> >> best regards >> >> steve >> >> > > Hi Steve, > > Thanks for the reply. We have: > > pickupexten = *8 ; Configure the pickup extension. Default > is *8 > > in features.conf. > > I will recompile on one of the sites this happens on. It's really odd, can > go for weeks without this happening and then a customer will report to me > that their extension is showing in use and I will login and there can be two > or three of these locks. On one site it actually makes asterisk impossible > to stop and I need to kill -9 > > We have stuck with version 1.4.22 as it has been so solid for us, no dumps > or deadlocks etc. We have tried to move to 1.4.25 and 1.4.29 but would > experience random weirdness that we just don't get with this version. > > When recompiled with this flag and if indeed it does show locks, what would > be the next step? > > Thanks for your help. > Brian > > > -- > _ > -- 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] AMI buffering event output?
Just wanted to say that the issue fixed itself with a linux kernel upgrade, in case anyone encounters this in the future and finds this thread. Did not have to recompile Asterisk to get it working on the new kernel. Regards, Örn 2011/6/2 Örn Arnarson : > en_US.UTF-8 in all cases. > > On Thu, Jun 2, 2011 at 3:33 PM, Mark Deneen wrote: >> 2011/6/2 Örn Arnarson : >>> To clarify; I observe the exact same results no matter how I connect >>> to the AMI on this particular server. I tried connecting FROM this >>> server to an AMI on another server to make sure it wasn't the telnet >>> client or some such, and then it worked perfectly. >>> >>> To answer the question, if I use the external IP address rather than >>> 127.0.0.1 I observe the same results. >>> >> >> echo $LANG >> on each server ? >> >> -- >> _ >> -- 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] AMI buffering event output?
en_US.UTF-8 in all cases. On Thu, Jun 2, 2011 at 3:33 PM, Mark Deneen wrote: > 2011/6/2 Örn Arnarson : >> To clarify; I observe the exact same results no matter how I connect >> to the AMI on this particular server. I tried connecting FROM this >> server to an AMI on another server to make sure it wasn't the telnet >> client or some such, and then it worked perfectly. >> >> To answer the question, if I use the external IP address rather than >> 127.0.0.1 I observe the same results. >> > > echo $LANG > on each server ? > > -- > _ > -- 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] AMI buffering event output?
To clarify; I observe the exact same results no matter how I connect to the AMI on this particular server. I tried connecting FROM this server to an AMI on another server to make sure it wasn't the telnet client or some such, and then it worked perfectly. To answer the question, if I use the external IP address rather than 127.0.0.1 I observe the same results. -- Örn On Thu, Jun 2, 2011 at 3:19 AM, Matt Riddell wrote: > On 1/06/11 11:03 PM, Örn Arnarson wrote: >> >> Hi Matt, >> >> Yes, passing two carriage returns. I login successfully. Here's >> example output (with my comments in []) >> >> Trying 127.0.0.1... >> Connected to localhost. >> Escape character is '^]'. >> Asterisk Call Manager/1.1 >> action: login >> username: phpagi >> secret: supersecretpassword >> events: on >> >> Response: Success >> Message: Authentication accepted > > It seems somewhat impossible that you would be getting different results > from different hosts. Are you using the same login? > > What if you use the external IP rather than 127.0.0.1 > > -- > Cheers, > > Matt Riddell > ___ > > http://www.venturevoip.com/news.php (Daily Asterisk News) > http://www.venturevoip.com/exchange.php (Full ITSP Solution) > http://www.venturevoip.com/cc.php (Call Centre Solutions) > > -- > _ > -- 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] AMI buffering event output?
Hi Alex, In the case of php, there's just an open socket. I'm using that php script on multiple other hosts with no buffering. The telnet client on the server is just "telnet", and I've tested telnetting to AMI on another server to make sure that the client isn't buffering with no problem. The problem only presents itself when telnetting to this particular AMI. Regards, Örn On Wed, Jun 1, 2011 at 11:02 AM, Alex Balashov wrote: > Are you using the same telnet client in both cases? > > On 06/01/2011 06:54 AM, Örn Arnarson wrote: > >> No, because the same is happening with telnet. If I telnet to AMI, I >> observe exactly the same behavior. Otherwise I would put it down to >> PHP. >> >> On Tue, May 31, 2011 at 5:41 PM, Alex Balashov >> wrote: >>> >>> On 05/31/2011 01:38 PM, Örn Arnarson wrote: >>>> >>>> Hi, >>>> >>>> I'm seeing weird behavior with AMI where no events are output until >>>> some input is detected (can be an empty line), at which time all the >>>> buffered output is spewed out at once. >>>> >>>> I am maintaining multiple Asterisk installations, and with one >>>> installation I have run into a weird buffering problem with AMI. >>>> The version is 1.6.1.11 in this particular case, which I am running at >>>> multiple locations, all without this problem. Additionally I have >>>> tried version 1.6.2.11 and 1.8.4.1, and the problem is consistent >>>> across these versions. >>>> >>>> manager.conf is identical across all these installations. >>>> >>>> The problem presents with a php script that opens a socket directly to >>>> AMI, a telnet client from the local machine to the AMI, but not when I >>>> telnet from the machine to a remote machine running AMI. >>>> >>>> Does anyone have any input as to what I can try? >>>> >>>> Best regards, >>>> Örn >>> >>> This is because of some blocking and/or read-write order issue in your >>> PHP >>> script. >>> >>> PHP is a web language, not for standalone scripts doing network I/O. >>> >>> -- >>> Alex Balashov - Principal >>> Evariste Systems LLC >>> 260 Peachtree Street NW >>> Suite 2200 >>> Atlanta, GA 30303 >>> Tel: +1-678-954-0670 >>> Fax: +1-404-961-1892 >>> Web: http://www.evaristesys.com/ >>> >>> -- >>> _ >>> -- 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 > > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > -- > _ > -- 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] AMI buffering event output?
Hi Matt, Yes, passing two carriage returns. I login successfully. Here's example output (with my comments in []) Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Asterisk Call Manager/1.1 action: login username: phpagi secret: supersecretpassword events: on Response: Success Message: Authentication accepted [No output after this point until I pass carriage return] [Now I press carriage return and two events appear, along with an error message for missing action] Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/2170 PeerStatus: Unregistered Cause: Expired Event: ExtensionStatus Privilege: call,all Exten: 2170 Context: default Hint: SIP/2170 Status: 4 Response: Error Message: Missing action in request On Tue, May 31, 2011 at 10:56 PM, Matt Riddell wrote: > On 1/06/11 5:38 AM, Örn Arnarson wrote: >> >> The problem presents with a php script that opens a socket directly to >> AMI, a telnet client from the local machine to the AMI, but not when I >> telnet from the machine to a remote machine running AMI. > > Are you 100% sure it's happening when you use telnet on the local machine? > > Are you passing two carriage returns after logging in? > > -- > Cheers, > > Matt Riddell > ___ > > http://www.venturevoip.com/news.php (Daily Asterisk News) > http://www.venturevoip.com/exchange.php (Full ITSP Solution) > http://www.venturevoip.com/cc.php (Call Centre Solutions) > > -- > _ > -- 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] AMI buffering event output?
No, because the same is happening with telnet. If I telnet to AMI, I observe exactly the same behavior. Otherwise I would put it down to PHP. On Tue, May 31, 2011 at 5:41 PM, Alex Balashov wrote: > On 05/31/2011 01:38 PM, Örn Arnarson wrote: >> >> Hi, >> >> I'm seeing weird behavior with AMI where no events are output until >> some input is detected (can be an empty line), at which time all the >> buffered output is spewed out at once. >> >> I am maintaining multiple Asterisk installations, and with one >> installation I have run into a weird buffering problem with AMI. >> The version is 1.6.1.11 in this particular case, which I am running at >> multiple locations, all without this problem. Additionally I have >> tried version 1.6.2.11 and 1.8.4.1, and the problem is consistent >> across these versions. >> >> manager.conf is identical across all these installations. >> >> The problem presents with a php script that opens a socket directly to >> AMI, a telnet client from the local machine to the AMI, but not when I >> telnet from the machine to a remote machine running AMI. >> >> Does anyone have any input as to what I can try? >> >> Best regards, >> Örn > > This is because of some blocking and/or read-write order issue in your PHP > script. > > PHP is a web language, not for standalone scripts doing network I/O. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI buffering event output?
Hi, I'm seeing weird behavior with AMI where no events are output until some input is detected (can be an empty line), at which time all the buffered output is spewed out at once. I am maintaining multiple Asterisk installations, and with one installation I have run into a weird buffering problem with AMI. The version is 1.6.1.11 in this particular case, which I am running at multiple locations, all without this problem. Additionally I have tried version 1.6.2.11 and 1.8.4.1, and the problem is consistent across these versions. manager.conf is identical across all these installations. The problem presents with a php script that opens a socket directly to AMI, a telnet client from the local machine to the AMI, but not when I telnet from the machine to a remote machine running AMI. Does anyone have any input as to what I can try? Best regards, Örn -- _ -- 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 and character sets and AMI
Hello again, Here's the header as it appears in 1.6.2.11 CLI output: INVITE sip:1...@192.168.10.169:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.3:5060;branch=z9hG4bK73713002;rport Max-Forwards: 70 From: "SIP ehf/Örn Arnarson" ;tag=as2813a8fe To: Contact: Here is the header with the same caller-id information in 1.8.0: INVITE sip:1...@192.168.10.169:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.3:5060;branch=z9hG4bK1ff411d7 Max-Forwards: 70 From: "SIP ehf/%C3%96rn Arnarson" ;tag=as701c8835 To: Contact: I have also attached PCAP files (from a different call setup, but with same information) for each scenario. Best regards, Örn On Fri, Oct 29, 2010 at 5:08 PM, Tilghman Lesher wrote: > On Friday 29 October 2010 11:46:01 Örn Arnarson wrote: >> Hi, >> >> Just tried upgrading to 1.8 and ran into two problem immediately; >> >> 1. Caller-ID behavior is different -- now when I set the caller-id >> name to something with special characters (Ö, for example), the SIP >> INVITE now has %C3%96 instead of the Ö character. I've tried doing >> Set(CALLERID(name-charset)=utf8) as well as iso8859-1, but it's always >> the same behavior. > > You'll need to include the relevant raw SIP messages for us to know if this > is compliant behavior or not. > >> 2. My AMI scripts have stopped working and Asterisk console shows a >> Broken Pipe error. Has the I/O to AMI changed? I had a quick glance >> through the change log and couldn't find anything indicating >> different. Haven't started looking at what the output looks like, but >> it would be nice if someone could point me to a document going through >> the changes so I don't have to re-invent the wheel. > > Any change to the protocol should be documented in UPGRADE.txt. AFAIK, > there has been no change to the actual protocol, but the various headers > may have changed slightly to ensure consistency between commands. > > -- > Tilghman Lesher > Digium, Inc. | Senior Software Developer > twitter: Corydon76 | IRC: Corydon76-dig (Freenode) > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- 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 1.6.2.11.pcap Description: application/extension-pcap 1.8.0.pcap Description: application/extension-pcap -- _ -- 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 and character sets and AMI
Hi, Just tried upgrading to 1.8 and ran into two problem immediately; 1. Caller-ID behavior is different -- now when I set the caller-id name to something with special characters (Ö, for example), the SIP INVITE now has %C3%96 instead of the Ö character. I've tried doing Set(CALLERID(name-charset)=utf8) as well as iso8859-1, but it's always the same behavior. 2. My AMI scripts have stopped working and Asterisk console shows a Broken Pipe error. Has the I/O to AMI changed? I had a quick glance through the change log and couldn't find anything indicating different. Haven't started looking at what the output looks like, but it would be nice if someone could point me to a document going through the changes so I don't have to re-invent the wheel. Anyone have any info on either one? Best regards, Örn Arnarson -- _ -- 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] Attended Transfer with REFER
Thanks a lot guys. Exactly what I needed. Best regards, Örn On Tue, Jan 26, 2010 at 8:48 PM, Olle E. Johansson wrote: > > 26 jan 2010 kl. 16.48 skrev Örn Arnarson: > > > Hi guys, > > > > I am wondering (and have been unable to find out thus far) whether > Asterisk sets some special channel variables or something when a call is > transfered with the REFER method. > > Basically, I'm trying to figure out if it is possible to somehow get a > transferred call back to the transferrer (as it is done with the built-in > atxfer) after X seconds (or an unsuccessful attempt). > > > > Using a timeout in the Dial command is not suitable unless I am able to > tell somehow that the call in question is being forwarded (which is of > course not the case, as the Dial command is called befer the REFER is sent). > > > > Can anyone think of a way to get the call back to the transferrer after > this timeout? > > > THe transferred call is sent to a context set with the channel variable > TRANSFER_CONTEXT before you call DIAL(). > > In there, run DUMPCHAN to see which variables you have and then dial with a > timeout. After the timeout, dial back. > > /O > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attended Transfer with REFER
Hi guys, I am wondering (and have been unable to find out thus far) whether Asterisk sets some special channel variables or something when a call is transfered with the REFER method. Basically, I'm trying to figure out if it is possible to somehow get a transferred call back to the transferrer (as it is done with the built-in atxfer) after X seconds (or an unsuccessful attempt). Using a timeout in the Dial command is not suitable unless I am able to tell somehow that the call in question is being forwarded (which is of course not the case, as the Dial command is called befer the REFER is sent). Can anyone think of a way to get the call back to the transferrer after this timeout? Best regards, Örn Arnarson -- _ -- 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] Languages
Hello, I found the respective functions in main/say.c -- now to find out if any language uses the same rules as Icelandic :-) Regards, Örn On Thu, Jan 14, 2010 at 3:40 PM, Danny Nicholas wrote: > AIR, French is set up to handle this, but I could very well be wrong; > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Örn Arnarson > *Sent:* Thursday, January 14, 2010 9:33 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] Languages > > > > Hello, > > > > What are the current methods for playing digits on different languages? I > presume the big ones like German have been dealt with, saying 2 and 20 to > announce 22. How is this currently decided? What about languages that say 20 > and 2? > > > > Is there a way of configuring via config files or recordings somehow? > Obviously you could record the sound file for 20 as "twenty and", but that > doesn't work when asterisk has to say the number 20. > > > > Is there a special case for every supported language in the source code? I > haven't been able to find where the distinction is made. > > > > Best regards, > > Örn > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Languages
Hello, What are the current methods for playing digits on different languages? I presume the big ones like German have been dealt with, saying 2 and 20 to announce 22. How is this currently decided? What about languages that say 20 and 2? Is there a way of configuring via config files or recordings somehow? Obviously you could record the sound file for 20 as "twenty and", but that doesn't work when asterisk has to say the number 20. Is there a special case for every supported language in the source code? I haven't been able to find where the distinction is made. Best regards, Örn -- _ -- 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] 1.6.1.10 Music On Hold
Brilliant, thanks a lot. Best regards, Örn On Tue, Nov 24, 2009 at 1:39 PM, Santiago Gimeno wrote: > Hi, > > I think it can be related to https://issues.asterisk.org/view.php?id=16268 > > Best regards, > > Santi > > 2009/11/24 Örn Arnarson > >> Hello again, >> >> I just tried version 1.6.1.9, and the MOH works well there. It seems to be >> a bug introduced in 1.6.1.10. >> >> Best regards, >> Örn >> >> 2009/11/23 Örn Arnarson >> >> Hello. >>> >>> I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On >>> Hold functionality has changed (or is bugged?). >>> >>> I have Aastra 6757i and Aastra 6731i phones, and now when i press the >>> MusicOnHold button / change lines on the phone, MOH no longer starts. It did >>> this in v 1.6.0.9. >>> >>> The invites received are exactly the same, only 1.6.1.10 doesn't ever >>> start MOH. >>> >>> Is there some configuration change I need to implement for this to work >>> properly? Was there a conscious change in Asterisk's behavior? >>> >>> Best regards, >>> Örn >>> >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1.10 Music On Hold
Hello again, I just tried version 1.6.1.9, and the MOH works well there. It seems to be a bug introduced in 1.6.1.10. Best regards, Örn 2009/11/23 Örn Arnarson > Hello. > > I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On > Hold functionality has changed (or is bugged?). > > I have Aastra 6757i and Aastra 6731i phones, and now when i press the > MusicOnHold button / change lines on the phone, MOH no longer starts. It did > this in v 1.6.0.9. > > The invites received are exactly the same, only 1.6.1.10 doesn't ever start > MOH. > > Is there some configuration change I need to implement for this to work > properly? Was there a conscious change in Asterisk's behavior? > > Best regards, > Örn > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6.1.10 Music On Hold
Hello. I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold functionality has changed (or is bugged?). I have Aastra 6757i and Aastra 6731i phones, and now when i press the MusicOnHold button / change lines on the phone, MOH no longer starts. It did this in v 1.6.0.9. The invites received are exactly the same, only 1.6.1.10 doesn't ever start MOH. Is there some configuration change I need to implement for this to work properly? Was there a conscious change in Asterisk's behavior? Best regards, Örn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk complaning about no such host -- never asked to contact the host it complains about
Hi, I'm seeing a very strange error when dealing with Diversions. If a call setup to a number comes to an Asterisk server, that server sends a request to a third proxy, that proxy sends the call back with a Diversion flag, Asterisk complains about the host not existing (and the host is the number). Here's the output from the Asterisk CLI with SIP debugging enabled: <--- SIP read from UDP://10.252.1.7:5060 ---> INVITE sip:...@10.252.1.20 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 10.252.1.7;branch=z9hG4bKfb6d.231a5066.0 Via: SIP/2.0/UDP 10.252.1.20:5060;received=10.252.1.20;branch=z9hG4bK02c3dba5;rport=5060 Max-Forwards: 69 From: "666" ;tag=as650133ff To: Contact: Call-ID: 48008f4b54d0913736bc3c0503a2c...@10.252.1.20 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.10 Date: Mon, 28 Sep 2009 15:41:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 284 Diversion: ;reason=unconditional v=0 o=root 1273543599 1273543599 IN IP4 10.252.1.7 s=Asterisk PBX 1.6.0.10 c=IN IP4 10.252.1.7 t=0 0 m=audio 39110 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes <-> --- (17 headers 13 lines) --- [Sep 28 15:41:24] WARNING[32316]: chan_sip.c:4224 create_addr: No such host: 555 Why would it be trying to contact the host 555? There's nothing in the invite indicating that as a host. Furthermore, the verbosity level was at the highest level, and I never saw the INVITE above come into the Dialplan. All I saw was the INVITE, and then the "No such host" error. Any ideas whatsoever? Best regards, Örn Arnarson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem in upgrading to 1.6.1.0
Hi D, Thanks for the suggestion. I put type = user in [general] of users.conf and that seems to have fixed it. Best regards, Örn On Mon, Sep 21, 2009 at 11:23 AM, D Tucny wrote: > In the 1.6.1.* branch the line type=peer seems to be required on each > user... > > d > > 2009/9/19 Örn Arnarson >> >> Sorry I wasn't more specific. >> >> The error message is just the standard 'Can't find that extension'. >> >> The problem is, however, that asterisk parses users.conf (and doesn't >> complain), but none of the users specified therein are loaded into the >> dialplan or even shown as peers (using sip show peers/users). A >> downgrade from 1.6.1.6 to 1.6.0.9 promptly fixed it, as with Oguzhan. >> >> Regards, >> Örn >> >> 2009/9/18 Benny Amorsen : >> > Örn Arnarson writes: >> > >> >> I'm seeing the same behavior in 1.6.1.6. >> >> >> >> Any info on this? >> > >> > It would be helpful if you copied the exact error message involving the >> > username field. >> > >> > >> > /Benny >> > >> > >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem in upgrading to 1.6.1.0
Sorry I wasn't more specific. The error message is just the standard 'Can't find that extension'. The problem is, however, that asterisk parses users.conf (and doesn't complain), but none of the users specified therein are loaded into the dialplan or even shown as peers (using sip show peers/users). A downgrade from 1.6.1.6 to 1.6.0.9 promptly fixed it, as with Oguzhan. Regards, Örn 2009/9/18 Benny Amorsen : > Örn Arnarson writes: > >> I'm seeing the same behavior in 1.6.1.6. >> >> Any info on this? > > It would be helpful if you copied the exact error message involving the > username field. > > > /Benny > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem in upgrading to 1.6.1.0
I'm seeing the same behavior in 1.6.1.6. Any info on this? On Wed, Apr 29, 2009 at 12:49 PM, Oguzhan Kayhan wrote: > Hello, > I just tried to upgrade to 1.6.1.0 from 1.6.0.9 and i had problems in > registering users. > As i see from debug it successfully reads from users.conf but later,when a > user tries to logon it say peer not found > And there were an error msg about mysql about the username field..Smthing > changed in mysql tables??? > > Now i downgraded to 1.6.0.9 again and everything is working.. > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UNIQUEID not the same in Dialplan aspassedtoAGI
Hi Danny, If 'both' is utilized instead of peer or self, the peer channel seems to be chosen (which was my original desired functionality). Best regards, Örn On Wed, Sep 9, 2009 at 2:49 PM, Danny Nicholas wrote: > I'll have to set this aside for a future experiment. I'm supposing that > both would mimic either the self or peer setting (although it seems possible > that it might generate 2 calls, the fork and the original). If you try it, > please post back for reference. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn Arnarson > Sent: Wednesday, September 09, 2009 9:40 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan > aspassedtoAGI > > Yes, exactly. > > I'm curious as to what would happen if I were to use both instead of peer. > :-) > > Regards, > Örn > > On Wed, Sep 9, 2009 at 2:17 PM, Danny Nicholas wrote: >> So when you do self, you get a "forked" call and peer runs the agi on the >> original call? >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn Arnarson >> Sent: Wednesday, September 09, 2009 9:13 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan >> aspassedtoAGI >> >> Hi Danny, >> >> Thanks again for your help. I figured out how to do this... It was >> through my own retardedness that I had messed it up. >> >> In the application map section of features.conf, I just swapped self for >> peer... >> used to be: >> tag1 => *1,self,AGI,tag1.agi >> >> is now: >> tag1 => *1,peer,AGI,tag1.agi >> >> That selects which channel is active for the call. I should have >> realized this earlier. >> >> Thanks again for your help. >> Örn >> >> 2009/9/9 Örn Arnarson : >>> Hi Danny, >>> >>> Thanks. Yes, that's where I'm getting the UNIQUEID. The problem is >>> that it is not for the same leg as the UNIQUEID in the Dialplan. If I >>> were able to get the same UNIQUEID somehow in both places, my problems >>> would be solved :-) >>> >>> Regards, >>> Örn >>> >>> On Wed, Sep 9, 2009 at 2:00 PM, Danny Nicholas wrote: >>>> Per this link - http://www.voip-info.org/wiki/view/Asterisk+AGI you >> should >>>> have a variable agi_uniqueid with the uniqueid of the leg available in >> the >>>> AGI. >>>> >>>> -Original Message- >>>> From: asterisk-users-boun...@lists.digium.com >>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn >> Arnarson >>>> Sent: Wednesday, September 09, 2009 8:52 AM >>>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>>> Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan as >>>> passedtoAGI >>>> >>>> If only features.conf accepted the normal syntax of running >>>> applications... As I understand it (and tested it), it must accept the >>>> application in the syntax name => keycode,WHO,application,parameters >>>> >>>> Goto would work, but then it will only be able to use Goto,priority >>>> >>>> On Wed, Sep 9, 2009 at 1:41 PM, Danny Nicholas wrote: >>>>> Actually (and this is probably an incorrect or misquoted statement), > any >>>>> action from features is a fork. If you change tag1 from >>>>> - tag1 => *1,self,AGI,tag1.agi >>>>> To >>>>> - tag1 => *1,self,Goto(runagi|s|1) >>>>> >>>>> The feature will jump to a context in your dialplan instead of directly >>>>> executing the AGI command. >>>>> >>>>> This may or may not work, but it should IMO. >>>>> >>>>> -Original Message- >>>>> From: asterisk-users-boun...@lists.digium.com >>>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn >> Arnarson >>>>> Sent: Wednesday, September 09, 2009 8:34 AM >>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>>>> Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan as >> passed >>>>> toAGI >>>>> >>>>> Thanks f
Re: [asterisk-users] UNIQUEID not the same in Dialplan aspassedtoAGI
Yes, exactly.
I'm curious as to what would happen if I were to use both instead of peer.
:-)
Regards,
Örn
On Wed, Sep 9, 2009 at 2:17 PM, Danny Nicholas wrote:
> So when you do self, you get a "forked" call and peer runs the agi on the
> original call?
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn Arnarson
> Sent: Wednesday, September 09, 2009 9:13 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan
> aspassedtoAGI
>
> Hi Danny,
>
> Thanks again for your help. I figured out how to do this... It was
> through my own retardedness that I had messed it up.
>
> In the application map section of features.conf, I just swapped self for
> peer...
> used to be:
> tag1 => *1,self,AGI,tag1.agi
>
> is now:
> tag1 => *1,peer,AGI,tag1.agi
>
> That selects which channel is active for the call. I should have
> realized this earlier.
>
> Thanks again for your help.
> Örn
>
> 2009/9/9 Örn Arnarson :
>> Hi Danny,
>>
>> Thanks. Yes, that's where I'm getting the UNIQUEID. The problem is
>> that it is not for the same leg as the UNIQUEID in the Dialplan. If I
>> were able to get the same UNIQUEID somehow in both places, my problems
>> would be solved :-)
>>
>> Regards,
>> Örn
>>
>> On Wed, Sep 9, 2009 at 2:00 PM, Danny Nicholas wrote:
>>> Per this link - http://www.voip-info.org/wiki/view/Asterisk+AGI you
> should
>>> have a variable agi_uniqueid with the uniqueid of the leg available in
> the
>>> AGI.
>>>
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com
>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn
> Arnarson
>>> Sent: Wednesday, September 09, 2009 8:52 AM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan as
>>> passedtoAGI
>>>
>>> If only features.conf accepted the normal syntax of running
>>> applications... As I understand it (and tested it), it must accept the
>>> application in the syntax name => keycode,WHO,application,parameters
>>>
>>> Goto would work, but then it will only be able to use Goto,priority
>>>
>>> On Wed, Sep 9, 2009 at 1:41 PM, Danny Nicholas wrote:
>>>> Actually (and this is probably an incorrect or misquoted statement), any
>>>> action from features is a fork. If you change tag1 from
>>>> - tag1 => *1,self,AGI,tag1.agi
>>>> To
>>>> - tag1 => *1,self,Goto(runagi|s|1)
>>>>
>>>> The feature will jump to a context in your dialplan instead of directly
>>>> executing the AGI command.
>>>>
>>>> This may or may not work, but it should IMO.
>>>>
>>>> -Original Message-
>>>> From: asterisk-users-boun...@lists.digium.com
>>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn
> Arnarson
>>>> Sent: Wednesday, September 09, 2009 8:34 AM
>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>> Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan as
> passed
>>>> toAGI
>>>>
>>>> Thanks for your reply.
>>>>
>>>> One thing I forgot to mention is that the AGI is called via
>>>> features.conf -- in a call, if you press, e.g. *1, you tag the call as
>>>> calltype 1.
>>>> As I cannot (as far as I know) call the AGI from features.conf with
>>>> parameters (such as ${UNIQUEID}), I need to get those parameters from
>>>> STDIN.
>>>>
>>>> This is an example from features.conf:
>>>> tag1 => *1,self,AGI,tag1.agi
>>>>
>>>> So, *1 in the active call calls this AGI. You might be right -- it's
>>>> just treated as another call (even though there is no fork, hangup or
>>>> transfer involved), but if this is the case, does anyone have an idea
>>>> how to pass parameters to tag1.agi via features.conf?
>>>>
>>>> Best regards,
>>>> Örn
>>>>
>>>> On Wed, Sep 9, 2009 at 1:14 PM, Danny Nicholas wrote:
>>>>> To quote Steve Edwards from an earlier post this month "The UniqueID
>>>>> consists of the origination time plus the number of calls for this
>>>> instance
>>>>
Re: [asterisk-users] UNIQUEID not the same in Dialplan as passedtoAGI
Hi Danny,
Thanks again for your help. I figured out how to do this... It was
through my own retardedness that I had messed it up.
In the application map section of features.conf, I just swapped self for peer...
used to be:
tag1 => *1,self,AGI,tag1.agi
is now:
tag1 => *1,peer,AGI,tag1.agi
That selects which channel is active for the call. I should have
realized this earlier.
Thanks again for your help.
Örn
2009/9/9 Örn Arnarson :
> Hi Danny,
>
> Thanks. Yes, that's where I'm getting the UNIQUEID. The problem is
> that it is not for the same leg as the UNIQUEID in the Dialplan. If I
> were able to get the same UNIQUEID somehow in both places, my problems
> would be solved :-)
>
> Regards,
> Örn
>
> On Wed, Sep 9, 2009 at 2:00 PM, Danny Nicholas wrote:
>> Per this link - http://www.voip-info.org/wiki/view/Asterisk+AGI you should
>> have a variable agi_uniqueid with the uniqueid of the leg available in the
>> AGI.
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn Arnarson
>> Sent: Wednesday, September 09, 2009 8:52 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan as
>> passedtoAGI
>>
>> If only features.conf accepted the normal syntax of running
>> applications... As I understand it (and tested it), it must accept the
>> application in the syntax name => keycode,WHO,application,parameters
>>
>> Goto would work, but then it will only be able to use Goto,priority
>>
>> On Wed, Sep 9, 2009 at 1:41 PM, Danny Nicholas wrote:
>>> Actually (and this is probably an incorrect or misquoted statement), any
>>> action from features is a fork. If you change tag1 from
>>> - tag1 => *1,self,AGI,tag1.agi
>>> To
>>> - tag1 => *1,self,Goto(runagi|s|1)
>>>
>>> The feature will jump to a context in your dialplan instead of directly
>>> executing the AGI command.
>>>
>>> This may or may not work, but it should IMO.
>>>
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com
>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn Arnarson
>>> Sent: Wednesday, September 09, 2009 8:34 AM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan as passed
>>> toAGI
>>>
>>> Thanks for your reply.
>>>
>>> One thing I forgot to mention is that the AGI is called via
>>> features.conf -- in a call, if you press, e.g. *1, you tag the call as
>>> calltype 1.
>>> As I cannot (as far as I know) call the AGI from features.conf with
>>> parameters (such as ${UNIQUEID}), I need to get those parameters from
>>> STDIN.
>>>
>>> This is an example from features.conf:
>>> tag1 => *1,self,AGI,tag1.agi
>>>
>>> So, *1 in the active call calls this AGI. You might be right -- it's
>>> just treated as another call (even though there is no fork, hangup or
>>> transfer involved), but if this is the case, does anyone have an idea
>>> how to pass parameters to tag1.agi via features.conf?
>>>
>>> Best regards,
>>> Örn
>>>
>>> On Wed, Sep 9, 2009 at 1:14 PM, Danny Nicholas wrote:
>>>> To quote Steve Edwards from an earlier post this month "The UniqueID
>>>> consists of the origination time plus the number of calls for this
>>> instance
>>>> of the Asterisk execution". Looking at the log you posted, these are two
>>>> separate (as far as Asterisk is concerned) calls. If you look in your
>>> CDR,
>>>> you should see an entry for each. If you want to track a call via an
>> AGI,
>>>> you should do this:
>>>> - exten => s,1,set(thisuid=${UNIQUEID})
>>>> - exten => s,2,AGI(youragi.agi|${thisuid})
>>>>
>>>> Just a guess, but UNIQUEID is probably reassigned on these events;
>>>> Hangup
>>>> Fork
>>>> Transfer
>>>> (go ahead guys, correct away - like I said, it's just a guess).
>>>>
>>>> -Original Message-
>>>> From: asterisk-users-boun...@lists.digium.com
>>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn
>> Arnarson
>>>> Sent: Wednesday, September 09, 2009 7:56 AM
>>>> To: Asterisk Users Mailing List - Non-Commercial Discuss
Re: [asterisk-users] UNIQUEID not the same in Dialplan as passedtoAGI
Hi Danny,
Thanks. Yes, that's where I'm getting the UNIQUEID. The problem is
that it is not for the same leg as the UNIQUEID in the Dialplan. If I
were able to get the same UNIQUEID somehow in both places, my problems
would be solved :-)
Regards,
Örn
On Wed, Sep 9, 2009 at 2:00 PM, Danny Nicholas wrote:
> Per this link - http://www.voip-info.org/wiki/view/Asterisk+AGI you should
> have a variable agi_uniqueid with the uniqueid of the leg available in the
> AGI.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn Arnarson
> Sent: Wednesday, September 09, 2009 8:52 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan as
> passedtoAGI
>
> If only features.conf accepted the normal syntax of running
> applications... As I understand it (and tested it), it must accept the
> application in the syntax name => keycode,WHO,application,parameters
>
> Goto would work, but then it will only be able to use Goto,priority
>
> On Wed, Sep 9, 2009 at 1:41 PM, Danny Nicholas wrote:
>> Actually (and this is probably an incorrect or misquoted statement), any
>> action from features is a fork. If you change tag1 from
>> - tag1 => *1,self,AGI,tag1.agi
>> To
>> - tag1 => *1,self,Goto(runagi|s|1)
>>
>> The feature will jump to a context in your dialplan instead of directly
>> executing the AGI command.
>>
>> This may or may not work, but it should IMO.
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn Arnarson
>> Sent: Wednesday, September 09, 2009 8:34 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan as passed
>> toAGI
>>
>> Thanks for your reply.
>>
>> One thing I forgot to mention is that the AGI is called via
>> features.conf -- in a call, if you press, e.g. *1, you tag the call as
>> calltype 1.
>> As I cannot (as far as I know) call the AGI from features.conf with
>> parameters (such as ${UNIQUEID}), I need to get those parameters from
>> STDIN.
>>
>> This is an example from features.conf:
>> tag1 => *1,self,AGI,tag1.agi
>>
>> So, *1 in the active call calls this AGI. You might be right -- it's
>> just treated as another call (even though there is no fork, hangup or
>> transfer involved), but if this is the case, does anyone have an idea
>> how to pass parameters to tag1.agi via features.conf?
>>
>> Best regards,
>> Örn
>>
>> On Wed, Sep 9, 2009 at 1:14 PM, Danny Nicholas wrote:
>>> To quote Steve Edwards from an earlier post this month "The UniqueID
>>> consists of the origination time plus the number of calls for this
>> instance
>>> of the Asterisk execution". Looking at the log you posted, these are two
>>> separate (as far as Asterisk is concerned) calls. If you look in your
>> CDR,
>>> you should see an entry for each. If you want to track a call via an
> AGI,
>>> you should do this:
>>> - exten => s,1,set(thisuid=${UNIQUEID})
>>> - exten => s,2,AGI(youragi.agi|${thisuid})
>>>
>>> Just a guess, but UNIQUEID is probably reassigned on these events;
>>> Hangup
>>> Fork
>>> Transfer
>>> (go ahead guys, correct away - like I said, it's just a guess).
>>>
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com
>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn
> Arnarson
>>> Sent: Wednesday, September 09, 2009 7:56 AM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: [asterisk-users] UNIQUEID not the same in Dialplan as passed to
>> AGI
>>>
>>> Hi,
>>>
>>> I've noticed that the UNIQUEID for a call is not the same in the
>>> Dialplan (when executed e.g. exten => s,n,NoOp(${UNIQUEID}) as it is
>>> when passed via STDIN to an AGI script.
>>> Is this normal, and is this supposed to behave this way?
>>>
>>> The UNIQUEID received in the AGI is usually .001 higher than the one
>>> in the dial plan -- but sometimes it is also a second behind.
>>> Here's an example from the dialplan, with the corresponding argument
>>> passed to the AGI:
>>>
>>> -- Executing [...@mac
Re: [asterisk-users] UNIQUEID not the same in Dialplan as passed toAGI
If only features.conf accepted the normal syntax of running
applications... As I understand it (and tested it), it must accept the
application in the syntax name => keycode,WHO,application,parameters
Goto would work, but then it will only be able to use Goto,priority
On Wed, Sep 9, 2009 at 1:41 PM, Danny Nicholas wrote:
> Actually (and this is probably an incorrect or misquoted statement), any
> action from features is a fork. If you change tag1 from
> - tag1 => *1,self,AGI,tag1.agi
> To
> - tag1 => *1,self,Goto(runagi|s|1)
>
> The feature will jump to a context in your dialplan instead of directly
> executing the AGI command.
>
> This may or may not work, but it should IMO.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn Arnarson
> Sent: Wednesday, September 09, 2009 8:34 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan as passed
> toAGI
>
> Thanks for your reply.
>
> One thing I forgot to mention is that the AGI is called via
> features.conf -- in a call, if you press, e.g. *1, you tag the call as
> calltype 1.
> As I cannot (as far as I know) call the AGI from features.conf with
> parameters (such as ${UNIQUEID}), I need to get those parameters from
> STDIN.
>
> This is an example from features.conf:
> tag1 => *1,self,AGI,tag1.agi
>
> So, *1 in the active call calls this AGI. You might be right -- it's
> just treated as another call (even though there is no fork, hangup or
> transfer involved), but if this is the case, does anyone have an idea
> how to pass parameters to tag1.agi via features.conf?
>
> Best regards,
> Örn
>
> On Wed, Sep 9, 2009 at 1:14 PM, Danny Nicholas wrote:
>> To quote Steve Edwards from an earlier post this month "The UniqueID
>> consists of the origination time plus the number of calls for this
> instance
>> of the Asterisk execution". Looking at the log you posted, these are two
>> separate (as far as Asterisk is concerned) calls. If you look in your
> CDR,
>> you should see an entry for each. If you want to track a call via an AGI,
>> you should do this:
>> - exten => s,1,set(thisuid=${UNIQUEID})
>> - exten => s,2,AGI(youragi.agi|${thisuid})
>>
>> Just a guess, but UNIQUEID is probably reassigned on these events;
>> Hangup
>> Fork
>> Transfer
>> (go ahead guys, correct away - like I said, it's just a guess).
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn Arnarson
>> Sent: Wednesday, September 09, 2009 7:56 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] UNIQUEID not the same in Dialplan as passed to
> AGI
>>
>> Hi,
>>
>> I've noticed that the UNIQUEID for a call is not the same in the
>> Dialplan (when executed e.g. exten => s,n,NoOp(${UNIQUEID}) as it is
>> when passed via STDIN to an AGI script.
>> Is this normal, and is this supposed to behave this way?
>>
>> The UNIQUEID received in the AGI is usually .001 higher than the one
>> in the dial plan -- but sometimes it is also a second behind.
>> Here's an example from the dialplan, with the corresponding argument
>> passed to the AGI:
>>
>> -- Executing [...@macro-internal-call:7]
>> NoOp("SIP/10.0.0.4-082a0658", "1252500374.334") in new stack
>>
>> agi_uniqueid: 1252500374.335
>>
>> And here's an example where the UNIQUEID is one second and one
>> fraction point behind:
>>
>> -- Executing [...@macro-internal-call:7]
>> NoOp("SIP/10.0.0.4-0825ef60", "1252500762.337") in new stack
>>
>> agi_uniqueid: 1252500763.338
>>
>> Any advice would be greatly appreciated. Can I use something else as a
>> unique identifier for a call? I'm trying to tag calls, but it is
>> proving difficult with the ever-changing UNIQUEID.
>>
>> I haven't found a rule as to when it is delayed by a second and when it
>> isn't.
>>
>> Best regards,
>> Örn
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo
Re: [asterisk-users] UNIQUEID not the same in Dialplan as passed to AGI
Hi again, and sorry for the spamming.
I've figured out now that Danny was indeed correct -- I'm looking at
two different legs here. On the one hand, I'm looking at the inbound
channel to Asterisk, whereas the AGI is dealing with the internal
channel to the endpoint.
Now, with that information, does anyone have a suggestion as to how I
can find out the UNIQUEID of the new leg? (Asterisk->Endpoint) in the
middle of the call? Should I be able to find it somehow through the
other call-leg, via the channel id or something?
Best regards,
Örn
2009/9/9 Örn Arnarson :
> Thanks for your reply.
>
> One thing I forgot to mention is that the AGI is called via
> features.conf -- in a call, if you press, e.g. *1, you tag the call as
> calltype 1.
> As I cannot (as far as I know) call the AGI from features.conf with
> parameters (such as ${UNIQUEID}), I need to get those parameters from
> STDIN.
>
> This is an example from features.conf:
> tag1 => *1,self,AGI,tag1.agi
>
> So, *1 in the active call calls this AGI. You might be right -- it's
> just treated as another call (even though there is no fork, hangup or
> transfer involved), but if this is the case, does anyone have an idea
> how to pass parameters to tag1.agi via features.conf?
>
> Best regards,
> Örn
>
> On Wed, Sep 9, 2009 at 1:14 PM, Danny Nicholas wrote:
>> To quote Steve Edwards from an earlier post this month "The UniqueID
>> consists of the origination time plus the number of calls for this instance
>> of the Asterisk execution". Looking at the log you posted, these are two
>> separate (as far as Asterisk is concerned) calls. If you look in your CDR,
>> you should see an entry for each. If you want to track a call via an AGI,
>> you should do this:
>> - exten => s,1,set(thisuid=${UNIQUEID})
>> - exten => s,2,AGI(youragi.agi|${thisuid})
>>
>> Just a guess, but UNIQUEID is probably reassigned on these events;
>> Hangup
>> Fork
>> Transfer
>> (go ahead guys, correct away - like I said, it's just a guess).
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn Arnarson
>> Sent: Wednesday, September 09, 2009 7:56 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] UNIQUEID not the same in Dialplan as passed to AGI
>>
>> Hi,
>>
>> I've noticed that the UNIQUEID for a call is not the same in the
>> Dialplan (when executed e.g. exten => s,n,NoOp(${UNIQUEID}) as it is
>> when passed via STDIN to an AGI script.
>> Is this normal, and is this supposed to behave this way?
>>
>> The UNIQUEID received in the AGI is usually .001 higher than the one
>> in the dial plan -- but sometimes it is also a second behind.
>> Here's an example from the dialplan, with the corresponding argument
>> passed to the AGI:
>>
>> -- Executing [...@macro-internal-call:7]
>> NoOp("SIP/10.0.0.4-082a0658", "1252500374.334") in new stack
>>
>> agi_uniqueid: 1252500374.335
>>
>> And here's an example where the UNIQUEID is one second and one
>> fraction point behind:
>>
>> -- Executing [...@macro-internal-call:7]
>> NoOp("SIP/10.0.0.4-0825ef60", "1252500762.337") in new stack
>>
>> agi_uniqueid: 1252500763.338
>>
>> Any advice would be greatly appreciated. Can I use something else as a
>> unique identifier for a call? I'm trying to tag calls, but it is
>> proving difficult with the ever-changing UNIQUEID.
>>
>> I haven't found a rule as to when it is delayed by a second and when it
>> isn't.
>>
>> Best regards,
>> Örn
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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Re: [asterisk-users] UNIQUEID not the same in Dialplan as passed to AGI
Thanks for your reply.
One thing I forgot to mention is that the AGI is called via
features.conf -- in a call, if you press, e.g. *1, you tag the call as
calltype 1.
As I cannot (as far as I know) call the AGI from features.conf with
parameters (such as ${UNIQUEID}), I need to get those parameters from
STDIN.
This is an example from features.conf:
tag1 => *1,self,AGI,tag1.agi
So, *1 in the active call calls this AGI. You might be right -- it's
just treated as another call (even though there is no fork, hangup or
transfer involved), but if this is the case, does anyone have an idea
how to pass parameters to tag1.agi via features.conf?
Best regards,
Örn
On Wed, Sep 9, 2009 at 1:14 PM, Danny Nicholas wrote:
> To quote Steve Edwards from an earlier post this month "The UniqueID
> consists of the origination time plus the number of calls for this instance
> of the Asterisk execution". Looking at the log you posted, these are two
> separate (as far as Asterisk is concerned) calls. If you look in your CDR,
> you should see an entry for each. If you want to track a call via an AGI,
> you should do this:
> - exten => s,1,set(thisuid=${UNIQUEID})
> - exten => s,2,AGI(youragi.agi|${thisuid})
>
> Just a guess, but UNIQUEID is probably reassigned on these events;
> Hangup
> Fork
> Transfer
> (go ahead guys, correct away - like I said, it's just a guess).
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn Arnarson
> Sent: Wednesday, September 09, 2009 7:56 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] UNIQUEID not the same in Dialplan as passed to AGI
>
> Hi,
>
> I've noticed that the UNIQUEID for a call is not the same in the
> Dialplan (when executed e.g. exten => s,n,NoOp(${UNIQUEID}) as it is
> when passed via STDIN to an AGI script.
> Is this normal, and is this supposed to behave this way?
>
> The UNIQUEID received in the AGI is usually .001 higher than the one
> in the dial plan -- but sometimes it is also a second behind.
> Here's an example from the dialplan, with the corresponding argument
> passed to the AGI:
>
> -- Executing [...@macro-internal-call:7]
> NoOp("SIP/10.0.0.4-082a0658", "1252500374.334") in new stack
>
> agi_uniqueid: 1252500374.335
>
> And here's an example where the UNIQUEID is one second and one
> fraction point behind:
>
> -- Executing [...@macro-internal-call:7]
> NoOp("SIP/10.0.0.4-0825ef60", "1252500762.337") in new stack
>
> agi_uniqueid: 1252500763.338
>
> Any advice would be greatly appreciated. Can I use something else as a
> unique identifier for a call? I'm trying to tag calls, but it is
> proving difficult with the ever-changing UNIQUEID.
>
> I haven't found a rule as to when it is delayed by a second and when it
> isn't.
>
> Best regards,
> Örn
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] UNIQUEID not the same in Dialplan as passed to AGI
Hi,
I've noticed that the UNIQUEID for a call is not the same in the
Dialplan (when executed e.g. exten => s,n,NoOp(${UNIQUEID}) as it is
when passed via STDIN to an AGI script.
Is this normal, and is this supposed to behave this way?
The UNIQUEID received in the AGI is usually .001 higher than the one
in the dial plan -- but sometimes it is also a second behind.
Here's an example from the dialplan, with the corresponding argument
passed to the AGI:
-- Executing [...@macro-internal-call:7]
NoOp("SIP/10.0.0.4-082a0658", "1252500374.334") in new stack
agi_uniqueid: 1252500374.335
And here's an example where the UNIQUEID is one second and one
fraction point behind:
-- Executing [...@macro-internal-call:7]
NoOp("SIP/10.0.0.4-0825ef60", "1252500762.337") in new stack
agi_uniqueid: 1252500763.338
Any advice would be greatly appreciated. Can I use something else as a
unique identifier for a call? I'm trying to tag calls, but it is
proving difficult with the ever-changing UNIQUEID.
I haven't found a rule as to when it is delayed by a second and when it isn't.
Best regards,
Örn
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Register Now: http://www.astricon.net
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Re: [asterisk-users] Broken Pipe error while using UpdateConfig command
I am seeing this problem on 1.6.0.1 when dialing a busy DAHDI channel... On Fri, Feb 13, 2009 at 8:40 AM, Rilawich Ango wrote: > I also experience that problem. Is it a bug? > > On Wed, Feb 4, 2009 at 5:53 AM, Mark Michelson > wrote: > > Remco Barendse wrote: > >> 1.4.23.1 is quite badly broken and there are no significant new > >> features > >> > > > > There are no new features at all, actually. What problems are you having > with > > 1.4.23.1? It doesn't accomplish much to say that it is "quite badly > broken" > > without at least telling what is wrong. > > > > We can't fix what's wrong if we don't know what's wrong to begin with. :) > > > > Mark Michelson > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with parking
Update on the matter:
I have reduced the config to one context and two peers:
extensions.conf:
[internal]
include => parkedcalls
exten => 2552,1,Dial(SIP/2552,,t)
exten => 2556,1,Dial(SIP/2556,,t)
sip.conf:
[general]
context=deadend
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
[2552]
type=friend
context=dialplan-1
host=dynamic
call-limit=10
defaultuser=2552
secret=xxx
[2556]
type=friend
context=dialplan-1
host=dynamic
call-limit=10
defaultuser=2556
secret=xxx
features.conf:
[general]
parkext => 10
parkpos => 11-14
parkingtime => 45
[featuremap]
blindxfer => #
That's all. I am starting to think that this must be an asterisk bug...
version is 1.6.0.1.
Regards,
Örn
On Thu, Feb 12, 2009 at 3:05 PM, Örn Arnarson wrote:
> Hi,
>
> I'm having problem with call parking.
> When I park call, either via transfer to xten or park digit sequence from
> features.conf, I hear the parking lot number read to me and the user gets
> transferred.
>
> However, MOH stops for the caller the moment user is transferred.
> The user can be retrieved by dialing the parked extension and voice
> resumes.
> If the parked user hangs up, the channel state does not update (nor the
> hint) and call seems to be live still.
> If the timeout for the park is reached, the user is not transferred back
> (nothing happens actually -- he is just able to stay parked forever).
> Even after the user hangs up, the parking lot extension exists and is
> callable, but when that is done (the parking extension is called), it
> answers and promptly hangs up and then is available for parking again.
>
> Yesterday I had everything working like a charm, and I don't think I have
> changed anything, although that seems increasingly unlikely since things
> don't usually break on their own. I've restarted asterisk numerous times and
> tried changing things I think are relevant, but with no avail.
>
> It does not matter whether the caller is coming in from a SIP trunk or a
> SIP peer.
>
> The SIP messages look normal. When the caller hangs up while parked he
> sends a BYE to the asterisk to the callee's number, and the asterisk replies
> with ACK.
>
> Any help would be appreciated.
>
> My comments in asterisk CLI output below prefixed by #
>
> Configuration files
>
> == features.conf ==
> [general]
> parkext => 10
> parkpos => 11-14
> parkinghints = yes
>
> == sip.conf ==
> [general]
> context=deadend
> allowoverlap=no
> bindport=5060
> bindaddr=0.0.0.0
> srvlookup=no
> subscribecontext=parkedcalls
> allowsubscribe=yes
> notifyringing=yes
> notifyhold=yes
> limitonpeers=yes
> canreinvite=no
>
> [Tal]
> context=incoming
> type=friend
> host=XXX.XXX.XXX.XXX
> canreinvite=no
> port=5060
> dtmfmode=rfc2833
> disallow=all
> allow=alaw
>
> [smg01]
> type=friend
> context=incoming
> host=XXX.XXX.XXX.XXX
> canreinvite=no
> port=5060
> dtmfmode=rfc2833
> disallow=all
> allow=alaw
>
> [2552]
> type=friend
> context=dialplan-1
> host=dynamic
> call-limit=10
> defaultuser=2552
> secret=xxx
>
> [2556]
> type=friend
> context=dialplan-1
> host=dynamic
> call-limit=10
> defaultuser=2556
> secret=xxx
>
> == extensions.conf ==
> [globals]
> CID => 5822550
> OCODE => 582
> XTENS => 255X
> DTRK => smg01
> CONF => 2559
> ADALNUMER => 2550
> ADALDIAL => SIP/2551&SIP/2552&SIP/2553&SIP/2554
> BAKVAKT => 7712555
>
> [general]
> static=yes
> writeprotect=yes
> clearglobalvars=no
> userscontext=default
>
> [dialplan-1]
> include => conferences
> include => ringgroups
> include => internal
> include => landlines
> include => gsm
> include => special
> include => international
> include => parkedcalls
>
> [dialplan-2]
> include => conferences
> include => ringgroups
> include => internal
> include => landlines
> include => parkedcalls
>
> [dialplan-3]
> include => conferences
> include => ringgroups
> include => internal
> include => landlines
> include => gsm
> include => parkedcalls
>
> [dialplan-4]
> include => conferences
> include => ringgroups
> include => internal
> include => landlines
> include => international
> include => parkedcalls
>
> [incoming]
> exten => 4891001,1,Set(CALLERID(name)=SMG01)
> exten => 4891001,n,Dial(${ADALDIAL})
> exten =>
> ${CID},1,Macro(ringgroup,${ADALDIAL},ADALNUMER,${ADALNUMER},ringgroup)
> exten => ${OCODE}${CONf},1,Goto(conferences,${EXTEN:3},1)
> exten => _${OCODE}${XTENS},1,Se
[asterisk-users] Problem with parking
Hi,
I'm having problem with call parking.
When I park call, either via transfer to xten or park digit sequence from
features.conf, I hear the parking lot number read to me and the user gets
transferred.
However, MOH stops for the caller the moment user is transferred.
The user can be retrieved by dialing the parked extension and voice resumes.
If the parked user hangs up, the channel state does not update (nor the
hint) and call seems to be live still.
If the timeout for the park is reached, the user is not transferred back
(nothing happens actually -- he is just able to stay parked forever).
Even after the user hangs up, the parking lot extension exists and is
callable, but when that is done (the parking extension is called), it
answers and promptly hangs up and then is available for parking again.
Yesterday I had everything working like a charm, and I don't think I have
changed anything, although that seems increasingly unlikely since things
don't usually break on their own. I've restarted asterisk numerous times and
tried changing things I think are relevant, but with no avail.
It does not matter whether the caller is coming in from a SIP trunk or a SIP
peer.
The SIP messages look normal. When the caller hangs up while parked he sends
a BYE to the asterisk to the callee's number, and the asterisk replies with
ACK.
Any help would be appreciated.
My comments in asterisk CLI output below prefixed by #
Configuration files
== features.conf ==
[general]
parkext => 10
parkpos => 11-14
parkinghints = yes
== sip.conf ==
[general]
context=deadend
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
subscribecontext=parkedcalls
allowsubscribe=yes
notifyringing=yes
notifyhold=yes
limitonpeers=yes
canreinvite=no
[Tal]
context=incoming
type=friend
host=XXX.XXX.XXX.XXX
canreinvite=no
port=5060
dtmfmode=rfc2833
disallow=all
allow=alaw
[smg01]
type=friend
context=incoming
host=XXX.XXX.XXX.XXX
canreinvite=no
port=5060
dtmfmode=rfc2833
disallow=all
allow=alaw
[2552]
type=friend
context=dialplan-1
host=dynamic
call-limit=10
defaultuser=2552
secret=xxx
[2556]
type=friend
context=dialplan-1
host=dynamic
call-limit=10
defaultuser=2556
secret=xxx
== extensions.conf ==
[globals]
CID => 5822550
OCODE => 582
XTENS => 255X
DTRK => smg01
CONF => 2559
ADALNUMER => 2550
ADALDIAL => SIP/2551&SIP/2552&SIP/2553&SIP/2554
BAKVAKT => 7712555
[general]
static=yes
writeprotect=yes
clearglobalvars=no
userscontext=default
[dialplan-1]
include => conferences
include => ringgroups
include => internal
include => landlines
include => gsm
include => special
include => international
include => parkedcalls
[dialplan-2]
include => conferences
include => ringgroups
include => internal
include => landlines
include => parkedcalls
[dialplan-3]
include => conferences
include => ringgroups
include => internal
include => landlines
include => gsm
include => parkedcalls
[dialplan-4]
include => conferences
include => ringgroups
include => internal
include => landlines
include => international
include => parkedcalls
[incoming]
exten => 4891001,1,Set(CALLERID(name)=SMG01)
exten => 4891001,n,Dial(${ADALDIAL})
exten =>
${CID},1,Macro(ringgroup,${ADALDIAL},ADALNUMER,${ADALNUMER},ringgroup)
exten => ${OCODE}${CONf},1,Goto(conferences,${EXTEN:3},1)
exten => _${OCODE}${XTENS},1,Set(CALLERID(name)=BEIN HRINGING)
exten => _${OCODE}${XTENS},n,Macro(internal-call,${EXTEN:3},incoming)
[internal]
exten => _${XTENS},1,Macro(internal-call,${EXTEN},internal)
exten => *80,1,Page(SIP/2553)
exten => *97,1,VoiceMailMain(${CALLERID(num)},s)
exten => *98,1,Playback(templokun)
exten => *99,1,Record(templokun.wav)
exten => _9X.,1,Macro(trunkdial,Tal,${EXTEN:1},${CALLERID(num)},landline)
[landlines]
exten =>
_800,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},tollfree)
exten =>
_[4-5]XX,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},landline)
exten => _177X,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},landline)
exten => 1817,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},landline)
exten => 1414,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},landline)
[gsm]
exten =>
_[6-8]XX,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},gsm)
[special]
exten => 112,1,Macro(trunkdial,${DTRK},${EXTEN},$CALLERID(num)},emergency)
exten => _11X,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},special)
exten => _15X,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},special)
exten =>
_9XX,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},special)
[international]
exten =>
_00x.,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},international)
[conferences]
exten => ${CONF},1,MeetMe(1,MsI)
[ringgroups]
exten =>
${ADALNUMER},1,Macro(ringgroup,${ADALDIAL},ADALNUMER,${ADALNUMER},internal)
[forwarding]
exten => _X.,1,Set(CALLERID(all)=${origcid})
exten => _X.,n,Set(GLOBAL(origcid)=)
exten => _X.,n,Goto(dialplan-1,${DB(CF/${EXTEN})},1)
[2550]
exten => s,1,Answer()
exten => s,n,Background(templokun)
exten
[asterisk-users] indications.conf entry for Iceland
Hi, Not sure where to submit this to so I'll try here. Below is the toneset for Iceland. Hopefully this can be added into the asterisk package. [is] description = Iceland ringcadence = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/5000 congestion = 425+250/250,0/250 callwaiting = 600/100,0/100,600/100,0/9000 record = 1400/500,0/15000 info = !950/330,!1400/330,!1800/330,0 stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400 Best regards, Örn ___ -- 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] CDR problems -- two call legs create only one CDR. Using ForkCDR() not even working.
Hello, When I bridge an incoming and outgoing call (attempting to simulate call-forwarding) I'm only getting one CDR -- that of the outgoing call. A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone on PSTN) and bridges the call. The only CDR created is from B to C. I have even tried using Answer() and ForkCDR() to get two CDRs, but to no avail. I am starting to wonder if there's a bug in the CDR generation in general, because I set up an extension to do only that: exten => 5822558,1,Answer() exten => 5822558,n,ForkCDR() exten => 5822558,n,Playback(tt-monkeys) exten => 5822558,n,Hangup() This is even given as an example on how to generate two CDRs from one call on this website: http://asterisk.name/asterisk/0596009623/asterisk-app-b-79.html I have been able to create two CDRs with the use of the Local/n channel, but the CDR is messy if I do so because I am required by law to change the caller-id for the outgoing call to that of the PBX, so both call legs seem to be originating from the Asterisk. Am I missing something? Any ideas appreciated. Best regards, Örn ___ -- 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] queue show name - callerID
There's not any direct way of which I am aware in a single command, but from
the shell you could do the following (and yes, this is a bit of a hack):
for i in `rasterisk -x "queue show" |grep wait |awk -F" " '{print $2}'`; do
rasterisk -x "core show channel $i" | grep "Caller ID";done
That will return (in order) the calls in your queue, their caller-ids and
caller id names. If the caller-id and caller-id-name are the same, each
entry will just be 2 repeating lines.
To skip the second entry (caller-id name), you can just add a colon to the
last grep command, like so:
for i in `rasterisk -x "queue show" |grep wait |awk -F" " '{print $2}'`; do
rasterisk -x "core show channel $i" | grep "Caller ID:";done
What this does, essentially, is check the callers' channels in the queue,
and then check each individual channel for the caller-id.
Best regards,
Örn
On Tue, Jul 1, 2008 at 7:20 PM, Marcin J. Kowalczyk
<[EMAIL PROTECTED]> wrote:
> Hi,
>
> Is there a way to show callerID of calls waiting in queue?
> queue show
> shows only channel not callerID
>
>
> Cheers,
> Marcin
>
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
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Register Now: http://www.astricon.net
asterisk-users mailing list
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Re: [asterisk-users] looking for trixbox manual in pdf
This isn't the official manual, but it's quite good. Helped me out when I started anyway. http://dumbme.voipeye.com.au/trixbox/trixbox_without_tears.pdf Best regards, Örn On Sun, May 4, 2008 at 9:07 AM, Sam Tam <[EMAIL PROTECTED]> wrote: > > I have been trying to source a trixbox ce manual in pdf but if anybody can > point me to the right direction then it would be good. > Sam > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ACD functionality , Skills for agents
I have often wondered the same thing. It seems to me to be random, or it works it out some way I am not familiar with. I have seen calls with wait time of 30 seconds get answered before calls with 30 minutes wait time from queues with equal weight. It would be great if someone who actually knows could answer or explain. Best regards, Örn Arnarson On Nov 21, 2007 2:15 PM, Kyriakos <[EMAIL PROTECTED]> wrote: > Guys can someone answer how the ACD works when it needs to decide which > call > to take next from queues with equal weights? Does it take the call with > the > longest period of watiting or does it work randomly? > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Kyriakos > Sent: Wednesday, November 21, 2007 11:08 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] ACD functionality , Skills for agents > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of James > FitzGibbon > Sent: Tuesday, November 20, 2007 6:03 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] ACD functionality , Skills for agents > > On Nov 20, 2007 10:16 AM, Kyriakos <[EMAIL PROTECTED]> wrote: > > > I have a question regarding ACD for queues. What happens when I have > 2 > > or more queues with same weight and each queue has a call in? How will > it > > decide which call will be routed to the next available agent? Will it > take > > the call with the longest waiting time in queue? If not how would I do > > this? > > Beware of queue weights. They have caused major problems in the past > for many people on this list. As I understand it, enabling weights > requires * to grab a lock on a large number of data structures related > to queue state, which can cause performance slowdowns and crashes. I > haven't seen reports of this recently, so it might be better in the > later 1.4 releases, but at one time it was a sure-fire recipe for > pain. > > > Also can someone point me to resources for making a single queue with > > customer calls tagged with agent skills? What I mean is instead of > having > > multiple queues Sales,Tech support, etc, have only a single queue with > > calls being tagged according to the customer's choice from IVR, so if a > > customer would choose SALES , the call would go into the queue with > other > > calls but it would only be answered from agents with the skill "SALES". > > This is something offered in other PBX systems like Avaya but im pretty > sure > > it can be done on Asterisk, right? > > It probably could be, but it would make reporting pretty difficult, as > the key fields in the queue log are the call id and the queue name. > While you could use the QueueLog() application to stick extra data > about the call (e.g the skill chosen from the IVR) into the queue log, > that would appear in one line only and require post-processing to glue > it together with the rest of the data for that call. I'm pretty sure > it wouldn't mesh nicely with the reporting package I use > (QueueMetrics). > > KM: I'm actually using the same package (Queuemetrics 1.4.2)! > > What I do for this is maintain queue (skill) membership in a database, > then add the channels to the appropriate queues when the agents log on > via a web page. Is there a particular reason you want to just have > one queue? > > KM: Well no if the ACD would work properly. As I mentioned there have > been > calls that were waiting in queue for 20 minutes because ACD was > distributing > calls from the rest of the queues with less waiting time. > > KM. > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nortel C15K <-> Asterisk
We have a CS2K and don't have problems with Asterisk communicating at all. The NGSS/SSTK on the CS2K doesn't have (as far as I know) support for user and password, but IP authentication is working fine. I don't know about the CS15K. We did run into some issues with getting phone calls to work properly after a software update on the switch, but that was resolved with a patch on the NGSS. [Nortel-SIP] type=friend port=5060 host=xx.xx.xx.xx dtmfmode=rfc2833 disallow=all context=from-trunk allow=alaw allow=ulaw allow=gsm Best regards, Örn On 10/26/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Has anyone had any luck getting an asterisk box to talk to a Nortel > C15K softswitch? Or any Nortel "sip" products? I've been playing with > it for several days and can't seem to pass calls either direction. I > know that whike the Nortel says the C15K speaks SIP, it really speaks > nortel's implementation of SIP, but I thought I could get it to at > least pass simple calls back and forth to an asterisk box. > > Right now, I can't even get asterisk to register with it. > Anyone have any ideas? > > thanks! > > > > register => username:[EMAIL PROTECTED] > > [nortel] > type=friend > fromuser=username > username=username > canreinvite=yes > secret=passwd > host= 192.168.1.20 > disallow=all > allow=gsm > allow=ulaw > allow=alaw > dtmfmode=rfc2833 > qualify=yes > nat=no > usereqphone=yes > context=from-nortel > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd one way RTP on SIP to SIP calls
Sorry for the spam, but there was a typo. I was running ISN09, but the upgrade was to ISN09u, which I am currently running. That was the upgrade that caused the interoperability problem with Asterisk that I mentioned. On 10/1/07, Örn Arnarson <[EMAIL PROTECTED]> wrote: > Good point. Here goes. > > I am running ISN09 (recently upgraded). Actually the upgrade caused a > lot of problems and now the CS2K has to be datafilled so that the > Asterisk trunks are Q764 and not Q767, lest the calls fail. > Additionally the NGSS/SST had to be patched up to date to fix another > issue. > > The NGSS config is pretty straight forward, no fancy options set. In > this version of * I had to change the following options to make it > work with this version of Asterisk: > Use OPTIONS for Heartbeat: No > Enforce CODEC-Compatibility: No (oddly enough, as the codecs are compatible) > Accepts Encapsulated ISUP: No > > sip.conf entry is like this: > [Nortel-SIP] > type=friend > host=1.1.1.1 > port=5060 > dtmfmode=rfc2833 > canreinvite=no > disallow=all > allow=alaw > allow=ulaw > context=default > > I think most of the other options were left at default, even though I > don't think that they are crucial. > > Best regards, > Örn > > On 10/1/07, Julio Arruda <[EMAIL PROTECTED]> wrote: > > > > Just a guess in fact..but.. > > I'm sure others would love to know how is the NGSS (SST now ?) config > > for this purpose, as well as your sip.conf and etc (one note, you are > > running SN09 or ISN09 ? > > Not sure, but this also would help others out there.. :-) > > > > > > > > Örn Arnarson wrote: > > > Julio, > > > > > > It seems you had something going there; I disallowed ISUP messages on > > > the SIP-T server and now I have two way audio. > > > > > > Thanks a lot for your help! > > > > > > Best regards, > > > Örn > > > > > > On 10/1/07, Örn Arnarson <[EMAIL PROTECTED]> wrote: > > >> You are right, the remote server is a SIP-T. > > >> > > >> I haven't had any problems connecting it to regular SIP servers > > >> thusfar though. Also like I mentioned, I don't have this one-way RTP > > >> problem with an earlier version of Asterisk. > > >> > > >> Thanks for your reply, > > >> Örn > > >> > > >> On 10/1/07, Julio Arruda <[EMAIL PROTECTED]> wrote: > > >>> Is this a SIP connection or a SIP-T one? Not sure (don't have access to > > >>> my previous life docs :-), but this seems to be a Session Server Trunks > > >>> doing SIP-T, not sure is the configuration you want...Have you tried to > > >>> contact their support ? > > >>> PS: this "c: application/ISUP;version=ANSI88;base=ANSI88", don't > > >>> remember seeing in plain SIP calls, so that is why I suspect is > > >>> configured as a SIP-T. > > >>> > > >>> Örn Arnarson wrote: > > >>>> Hi everyone, > > >>>> > > >>>> I'm having an odd problem with one way RTP on SIP to SIP calls. > > >>>> I have two SIP servers, one is an Asterisk and the remote SIP server > > >>>> is a Nortel SIP server. > > >>>> > > >>>> When a call comes to the Nortel server through the PSTN and is routed > > >>>> to the Asterisk, audio is fine. Two way RTP and no problems. When a > > >>>> SIP client registered on the Nortel server calls the Asterisk, the > > >>>> Asterisk doesn't seem to send any RTP. > > >>>> > > >>>> As far as I can tell, there isn't anything wrong with the call setup. > > >>>> > > >>>> show core version shows: > > >>>> Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on > > >>>> 2007-05-17 06:39:34 UTC > > >>>> > > >>>> SIP and RTP debugging on Asterisk shows this: > > >>>> http://www.arnarson.net/~orn/calldebug.txt > > >>>> > > >>>> On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by > > >>>> root @ build.trixbox.org on a i686 running Linux on 2007-04-25 > > >>>> 19:59:21 UTC) on the same network (same subnet and physical location) > > >>>> as the 1.4.4 this problem does not exist. There is no RTP problem when > > >>>> SIP clients registered on Nortel call. > > >>>> > > >>>> If anyone could help or suggest anything it would be greatly > > >>>> appreciated. > > >>>> > > >>>> Best regards, > > >>>> Örn > > >>>> ___ > > > > > > ___ > > > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd one way RTP on SIP to SIP calls
Good point. Here goes. I am running ISN09 (recently upgraded). Actually the upgrade caused a lot of problems and now the CS2K has to be datafilled so that the Asterisk trunks are Q764 and not Q767, lest the calls fail. Additionally the NGSS/SST had to be patched up to date to fix another issue. The NGSS config is pretty straight forward, no fancy options set. In this version of * I had to change the following options to make it work with this version of Asterisk: Use OPTIONS for Heartbeat: No Enforce CODEC-Compatibility: No (oddly enough, as the codecs are compatible) Accepts Encapsulated ISUP: No sip.conf entry is like this: [Nortel-SIP] type=friend host=1.1.1.1 port=5060 dtmfmode=rfc2833 canreinvite=no disallow=all allow=alaw allow=ulaw context=default I think most of the other options were left at default, even though I don't think that they are crucial. Best regards, Örn On 10/1/07, Julio Arruda <[EMAIL PROTECTED]> wrote: > > Just a guess in fact..but.. > I'm sure others would love to know how is the NGSS (SST now ?) config > for this purpose, as well as your sip.conf and etc (one note, you are > running SN09 or ISN09 ? > Not sure, but this also would help others out there.. :-) > > > > Örn Arnarson wrote: > > Julio, > > > > It seems you had something going there; I disallowed ISUP messages on > > the SIP-T server and now I have two way audio. > > > > Thanks a lot for your help! > > > > Best regards, > > Örn > > > > On 10/1/07, Örn Arnarson <[EMAIL PROTECTED]> wrote: > >> You are right, the remote server is a SIP-T. > >> > >> I haven't had any problems connecting it to regular SIP servers > >> thusfar though. Also like I mentioned, I don't have this one-way RTP > >> problem with an earlier version of Asterisk. > >> > >> Thanks for your reply, > >> Örn > >> > >> On 10/1/07, Julio Arruda <[EMAIL PROTECTED]> wrote: > >>> Is this a SIP connection or a SIP-T one? Not sure (don't have access to > >>> my previous life docs :-), but this seems to be a Session Server Trunks > >>> doing SIP-T, not sure is the configuration you want...Have you tried to > >>> contact their support ? > >>> PS: this "c: application/ISUP;version=ANSI88;base=ANSI88", don't > >>> remember seeing in plain SIP calls, so that is why I suspect is > >>> configured as a SIP-T. > >>> > >>> Örn Arnarson wrote: > >>>> Hi everyone, > >>>> > >>>> I'm having an odd problem with one way RTP on SIP to SIP calls. > >>>> I have two SIP servers, one is an Asterisk and the remote SIP server > >>>> is a Nortel SIP server. > >>>> > >>>> When a call comes to the Nortel server through the PSTN and is routed > >>>> to the Asterisk, audio is fine. Two way RTP and no problems. When a > >>>> SIP client registered on the Nortel server calls the Asterisk, the > >>>> Asterisk doesn't seem to send any RTP. > >>>> > >>>> As far as I can tell, there isn't anything wrong with the call setup. > >>>> > >>>> show core version shows: > >>>> Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on > >>>> 2007-05-17 06:39:34 UTC > >>>> > >>>> SIP and RTP debugging on Asterisk shows this: > >>>> http://www.arnarson.net/~orn/calldebug.txt > >>>> > >>>> On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by > >>>> root @ build.trixbox.org on a i686 running Linux on 2007-04-25 > >>>> 19:59:21 UTC) on the same network (same subnet and physical location) > >>>> as the 1.4.4 this problem does not exist. There is no RTP problem when > >>>> SIP clients registered on Nortel call. > >>>> > >>>> If anyone could help or suggest anything it would be greatly appreciated. > >>>> > >>>> Best regards, > >>>> Örn > >>>> ___ > > > ___ > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd one way RTP on SIP to SIP calls
Julio, It seems you had something going there; I disallowed ISUP messages on the SIP-T server and now I have two way audio. Thanks a lot for your help! Best regards, Örn On 10/1/07, Örn Arnarson <[EMAIL PROTECTED]> wrote: > You are right, the remote server is a SIP-T. > > I haven't had any problems connecting it to regular SIP servers > thusfar though. Also like I mentioned, I don't have this one-way RTP > problem with an earlier version of Asterisk. > > Thanks for your reply, > Örn > > On 10/1/07, Julio Arruda <[EMAIL PROTECTED]> wrote: > > Is this a SIP connection or a SIP-T one? Not sure (don't have access to > > my previous life docs :-), but this seems to be a Session Server Trunks > > doing SIP-T, not sure is the configuration you want...Have you tried to > > contact their support ? > > PS: this "c: application/ISUP;version=ANSI88;base=ANSI88", don't > > remember seeing in plain SIP calls, so that is why I suspect is > > configured as a SIP-T. > > > > Örn Arnarson wrote: > > > Hi everyone, > > > > > > I'm having an odd problem with one way RTP on SIP to SIP calls. > > > I have two SIP servers, one is an Asterisk and the remote SIP server > > > is a Nortel SIP server. > > > > > > When a call comes to the Nortel server through the PSTN and is routed > > > to the Asterisk, audio is fine. Two way RTP and no problems. When a > > > SIP client registered on the Nortel server calls the Asterisk, the > > > Asterisk doesn't seem to send any RTP. > > > > > > As far as I can tell, there isn't anything wrong with the call setup. > > > > > > show core version shows: > > > Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on > > > 2007-05-17 06:39:34 UTC > > > > > > SIP and RTP debugging on Asterisk shows this: > > > http://www.arnarson.net/~orn/calldebug.txt > > > > > > On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by > > > root @ build.trixbox.org on a i686 running Linux on 2007-04-25 > > > 19:59:21 UTC) on the same network (same subnet and physical location) > > > as the 1.4.4 this problem does not exist. There is no RTP problem when > > > SIP clients registered on Nortel call. > > > > > > If anyone could help or suggest anything it would be greatly appreciated. > > > > > > Best regards, > > > Örn > > > ___ > > > > > > Sign up now for AstriCon 2007! September 25-28th. > > > http://www.astricon.net/ > > > > > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd one way RTP on SIP to SIP calls
You are right, the remote server is a SIP-T. I haven't had any problems connecting it to regular SIP servers thusfar though. Also like I mentioned, I don't have this one-way RTP problem with an earlier version of Asterisk. Thanks for your reply, Örn On 10/1/07, Julio Arruda <[EMAIL PROTECTED]> wrote: > Is this a SIP connection or a SIP-T one? Not sure (don't have access to > my previous life docs :-), but this seems to be a Session Server Trunks > doing SIP-T, not sure is the configuration you want...Have you tried to > contact their support ? > PS: this "c: application/ISUP;version=ANSI88;base=ANSI88", don't > remember seeing in plain SIP calls, so that is why I suspect is > configured as a SIP-T. > > Örn Arnarson wrote: > > Hi everyone, > > > > I'm having an odd problem with one way RTP on SIP to SIP calls. > > I have two SIP servers, one is an Asterisk and the remote SIP server > > is a Nortel SIP server. > > > > When a call comes to the Nortel server through the PSTN and is routed > > to the Asterisk, audio is fine. Two way RTP and no problems. When a > > SIP client registered on the Nortel server calls the Asterisk, the > > Asterisk doesn't seem to send any RTP. > > > > As far as I can tell, there isn't anything wrong with the call setup. > > > > show core version shows: > > Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on > > 2007-05-17 06:39:34 UTC > > > > SIP and RTP debugging on Asterisk shows this: > > http://www.arnarson.net/~orn/calldebug.txt > > > > On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by > > root @ build.trixbox.org on a i686 running Linux on 2007-04-25 > > 19:59:21 UTC) on the same network (same subnet and physical location) > > as the 1.4.4 this problem does not exist. There is no RTP problem when > > SIP clients registered on Nortel call. > > > > If anyone could help or suggest anything it would be greatly appreciated. > > > > Best regards, > > Örn > > ___ > > > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd one way RTP on SIP to SIP calls
Hi everyone, I'm having an odd problem with one way RTP on SIP to SIP calls. I have two SIP servers, one is an Asterisk and the remote SIP server is a Nortel SIP server. When a call comes to the Nortel server through the PSTN and is routed to the Asterisk, audio is fine. Two way RTP and no problems. When a SIP client registered on the Nortel server calls the Asterisk, the Asterisk doesn't seem to send any RTP. As far as I can tell, there isn't anything wrong with the call setup. show core version shows: Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on 2007-05-17 06:39:34 UTC SIP and RTP debugging on Asterisk shows this: http://www.arnarson.net/~orn/calldebug.txt On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by root @ build.trixbox.org on a i686 running Linux on 2007-04-25 19:59:21 UTC) on the same network (same subnet and physical location) as the 1.4.4 this problem does not exist. There is no RTP problem when SIP clients registered on Nortel call. If anyone could help or suggest anything it would be greatly appreciated. Best regards, Örn ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
