[asterisk-users] OT - Which Sugarcm 6.2 plugin to add click-to-dial (with asterisk) ?
Hi, Which free or non-free (as beer) Sugarcrm plugin would you recommend to add click to dial feature with asterisk ? I can see a quite long list of such plugins but not all of them seem up-to-date (judging by comparing with latest Sugarcrm version number). Regards. -- _ -- 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] OT - Which Sugarcm 6.2 plugin to add click-to-dial (with asterisk) ?
On 04/16/2012 08:06 AM, Olivier wrote: Hi, Which free or non-free (as beer) Sugarcrm plugin would you recommend to add click to dial feature with asterisk ? I can see a quite long list of such plugins but not all of them seem up-to-date (judging by comparing with latest Sugarcrm version number). I would appreciate some feedback (paid or non-paid) from the Community as well. I looked at this last week and it seems that there are only 2 that may be still actively developed and might work (but I have tried neither!): SugarDir and Yaai. The first one can be found at the SugarCRM website (SugarForge?) and the last one can be found here: https://github.com/blak3r/yaai If you find something that works please let us know. Regards, Patrick -- _ -- 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] Pickup calls coming from queues
-Original Message- From: Niccolò Belli [mailto:darkba...@linuxsystems.it] Sent: Monday, 16 April 2012 4:21 a.m. To: asterisk-users@lists.digium.com Cc: siva...@paradise.net.nz Subject: Re: [asterisk-users] Pickup calls coming from queues Il 20/01/2012 20:32, Alec Davis ha scritto: This maybe not what you want. Our solution was monitor a queue with a BLF, instead of a queue member This reviewhttps://reviewboard.asterisk.org/r/1619/ allows a BLF lamp to flash when a queue is ringing, then the queue can be picked up by the BLF button. The hint does work very well but I really didn't understand how you did pickup the call in example 2... I don't have static members and each (dynamic) member may be logged to another queue too. So even if I know SIP/155 is a (dynamic) member of Queue1 I can't pick up 155 if it's ringing because of another call coming from Queue2. Thanks, Niccolò The trick is, don't try to pickup the ringing device (SIP/155), pickup the queue's extension, in our example Pickup(itg@trusted). We too, have some users logged into 3 queues at the same time, with more than 3 others only watching one of the queues. [ivr-dialextension] (1) exten = 8501,1,Goto(itg-queue,itg,1) ;Jump to the ITG queue context [itg-queue] (2) exten = itg,1,Queue(itg_queue,crhH,,,127) [trusted] (3) exten = 8501,hint,Queue:itg_queue;Provide a hint for the queue exten = _**8501,1,Pickup(itg@trusted);Pickup the queue With the above 3 contexts; (1) the caller finally has dialled 8501 from the IVR (2) Which jumps to the itg-queue context and rings the phones that are dynamically logged in. (3) all phones start in the 'trusted' context, and dialling **8501 will pickup the ringing extension. I just dialled into our IVR, and the execution path was as follows (1) -- Executing [8501@ivr-dialextension:1] Goto(DAHDI/i1/214X-cea, itg-queue,itg,1) in new stack -- Goto (itg-queue,itg,1) (2) -- Executing [itg@itg-queue:1] Queue(DAHDI/i1/214X-cea, itg_queue,crhH,,,127) in new stack [2012-04-16 19:16:28.699] NOTICE[24023]: app_queue.c:2516 join_queue: ALEC queue=itg_queue count++ =1 == Extension Changed 8501[trusted] new state Ringing for Notify User GXP0001 == Using SIP RTP CoS mark 5 -- ALEC Queue trying itg@itg-queue == Extension Changed 8512[trusted] new state Ringing for Notify User GXP0001 -- SIP/GXP0001-185f is ringing (3) Then the pickup, I'm not there but dialling **8501 would pickup the ringing phone GXP0001 Hope this better explains it. Alec Davis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNOM phones? Please test this patch (broken hints with notifycid=yes)
Patch: https://issues.asterisk.org/jira/secure/attachment/42605/local_remote_hint2.diff https://issues.asterisk.org/jira/browse/ASTERISK-16735?focusedCommentId=191720page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-191720 It's already merged in asterisk 10.4-rc1, it breaks hints for me but I suspect it may be a snom's bug. Cheers, Niccolò -- _ -- 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] Pickup calls coming from queues
I suspected it, but it didn't work at first. I fear I didn't understand what the context refers to in Pickup(extension[@context]). I will make an example: phone-100 wants to pick up a ringing phone-200 (call comes from my-sip-provider). This is my sip.conf [phone-100] context=context-100 [phone-200] context=context-200 [my-sip-provider] context=from-my-sip-provider This is my extensions.conf [context-100] exten = test,hint,Queue:MyQueue exten = test,1,Pickup(myphonenumber@from-my-sip-provider) [...] [context-200] [...] [from-my-sip-provider] exten = myphonenumber,1,Queue(MyQueue,r) same = n,Hangup() I expected to use from-my-sip-provider as context in Pickup, unfortunately it didn't work. So I tried both context-100 and context-200 as context in Pickup and they *both* worked! What's the logic behind Pickup's context? Thanks, Niccolò -- _ -- 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] Set variables from one asterisk ta a second.
Hi,
If you are using IAX and a later version (I know it works in 1.8.x) you
can use IAXVAR.
The following URL has a post which has a good example.
http://lists.digium.com/pipermail/asterisk-dev/2006-August/022313.html
Kind Regards
Stuart Elvish
On 04/16/2012 08:16 AM, Steve Edwards wrote:
On Sun, 15 Apr 2012, Olivier CALVANO wrote:
actually, i have a asterisk server with all SIP Account.
this Asterisk server sent all outgoing call to a second Asterisk
server (and this asterisk sent to the
telco)
On the first Asterisk, i use:
exten = _x,1,Set(CDR(CodeTier)=BUS-FRAMOBI)
exten = _x,2,Set(CALLERID(num)=${CALLERIDNUM})
exten = _x,3,Dial(IAX2/Srv2/${EXTEN},180,rt)
exten = _x,4,Hangup
i have SIP user: USRSIP001
(user sip is in realtime)
he use this name with a password
i want that the first server sent to the second into a variable the
USRSIP001
for get it into a AGI script.
It's possible ?
Yes.
(I'm just a 1.2 Luddite, so the exact capabilities and syntax available
to your version may be different.)
The first question is 'Do you want to use SIP or IAX?' You've used IAX
in your dialplan snippet, but you may want to consider SIP. The initial
configuration is a bit more involved, but you will have a more flexible
and maintainable solution.
Using IAX is simpler but you are limited to 'overloading' the caller ID
'name' and 'num' fields*. If you have more than a couple of fields of
data to pass you may find it easier to pass a 'key' (like the server
name and the channel unique ID) and use that to retrieve data from a
database instead of having to parse a bunch of fields from the caller ID
name or number.
Using SIP you can also pass data by adding custom SIP headers.
Personally, I've always used IAX because it was easy and it worked in my
environment. If I were to start over, I would seriously consider SIP.
A simple IAX example snippet...
On server1:
exten = *,n,set(CALLERID(name)=olivier-calvano)
exten = *,n,dial(iax2/server2/${EXTEN})
exten = *,n,hangup()
On server2:
exten = *,n,set(FIRST=${CUT(CALLERID(name),,1)})
exten = *,n,set(LAST=${CUT(CALLERID(name),,2)})
exten = *,n,agi(lookup-client,${FIRST},${LAST})
exten = *,n,hangup()
*) The extension and context are also under your control and can be set
in the IAX 'dial string' but manipulating these fields to pass multiple
data fields can get convoluted.
--
_
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Re: [asterisk-users] Caller ID problem
Thanks Danny. I test it with blind transfer and hey, you're right, the caller ID passed successfully, but the attended transfer doesn't. What version did you refer to by saying 10.x ? Asterisk? Shoudn't current version of asterisk is 1.x and should move to 2.x instead of a big jump to 10.x ? Thanks :) BR, Anam Totally newbie On 4/16/12, Danny Nicholas da...@debsinc.com wrote: Do a blind transfer instead of attended transfer - the under the covers changes in 10.X handle this for attended transfers, but to the best of my knowledge, the blind transfer is the only solution in the 1.X tree. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Sunday, April 15, 2012 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Caller ID problem Hi, I'm running asterisk 1.8.7.0 FreePBX 2.8.1 IP Phone Yealink T20 Trustrpid and sendrpid is on the sip.conf Let say I pickup a call on ext A using *8, the caller's caller ID successfully passed to my phone. I decide to pass the call to ext B. On phone B, it display ext A not the original's caller ID. I want on phone B it display the caller's caller ID. Is there any solution for this? I already googling this for around a week but found no solution yet :( Thanks and BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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] Caller ID problem
Hi Anam, Hope this helps explain Asterisk version numbering: http://leifmadsen.wordpress.com/2011/08/29/asterisk-10-asterisk-1-hh10/ Easy to get confused!. Cheers, AJ. - Original Message - From: Satria Anamarta anam.satri...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 16 April, 2012 12:10:27 PM Subject: Re: [asterisk-users] Caller ID problem Thanks Danny. I test it with blind transfer and hey, you're right, the caller ID passed successfully, but the attended transfer doesn't. What version did you refer to by saying 10.x ? Asterisk? Shoudn't current version of asterisk is 1.x and should move to 2.x instead of a big jump to 10.x ? Thanks :) BR, Anam Totally newbie On 4/16/12, Danny Nicholas da...@debsinc.com wrote: Do a blind transfer instead of attended transfer - the under the covers changes in 10.X handle this for attended transfers, but to the best of my knowledge, the blind transfer is the only solution in the 1.X tree. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Sunday, April 15, 2012 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Caller ID problem Hi, I'm running asterisk 1.8.7.0 FreePBX 2.8.1 IP Phone Yealink T20 Trustrpid and sendrpid is on the sip.conf Let say I pickup a call on ext A using *8, the caller's caller ID successfully passed to my phone. I decide to pass the call to ext B. On phone B, it display ext A not the original's caller ID. I want on phone B it display the caller's caller ID. Is there any solution for this? I already googling this for around a week but found no solution yet :( Thanks and BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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] Set variables from one asterisk ta a second.
Hi
greats thanks that work very good
Olivier
Le 16 avril 2012 12:47, Stuart Elvish - IP Exchange Systems
asterisk.li...@ipesys.com a écrit :
Hi,
If you are using IAX and a later version (I know it works in 1.8.x) you
can use IAXVAR.
The following URL has a post which has a good example.
http://lists.digium.com/pipermail/asterisk-dev/2006-August/022313.html
Kind Regards
Stuart Elvish
On 04/16/2012 08:16 AM, Steve Edwards wrote:
On Sun, 15 Apr 2012, Olivier CALVANO wrote:
actually, i have a asterisk server with all SIP Account.
this Asterisk server sent all outgoing call to a second Asterisk
server (and this asterisk sent to the
telco)
On the first Asterisk, i use:
exten = _x,1,Set(CDR(CodeTier)=BUS-FRAMOBI)
exten = _x,2,Set(CALLERID(num)=${CALLERIDNUM})
exten = _x,3,Dial(IAX2/Srv2/${EXTEN},180,rt)
exten = _x,4,Hangup
i have SIP user: USRSIP001
(user sip is in realtime)
he use this name with a password
i want that the first server sent to the second into a variable the
USRSIP001
for get it into a AGI script.
It's possible ?
Yes.
(I'm just a 1.2 Luddite, so the exact capabilities and syntax available
to your version may be different.)
The first question is 'Do you want to use SIP or IAX?' You've used IAX
in your dialplan snippet, but you may want to consider SIP. The initial
configuration is a bit more involved, but you will have a more flexible
and maintainable solution.
Using IAX is simpler but you are limited to 'overloading' the caller ID
'name' and 'num' fields*. If you have more than a couple of fields of
data to pass you may find it easier to pass a 'key' (like the server
name and the channel unique ID) and use that to retrieve data from a
database instead of having to parse a bunch of fields from the caller ID
name or number.
Using SIP you can also pass data by adding custom SIP headers.
Personally, I've always used IAX because it was easy and it worked in my
environment. If I were to start over, I would seriously consider SIP.
A simple IAX example snippet...
On server1:
exten = *,n, set(CALLERID(name)=olivier-calvano)
exten = *,n, dial(iax2/server2/${EXTEN})
exten = *,n, hangup()
On server2:
exten = *,n, set(FIRST=${CUT(CALLERID(name),,1)})
exten = *,n, set(LAST=${CUT(CALLERID(name),,2)})
exten = *,n, agi(lookup-client,${FIRST},${LAST})
exten = *,n, hangup()
*) The extension and context are also under your control and can be set
in the IAX 'dial string' but manipulating these fields to pass multiple
data fields can get convoluted.
--
_
-- 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:
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--
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Re: [asterisk-users] Unable to access the running directory (Permission denied).
On Sat, Apr 07, 2012 at 08:20:56PM +, Noah Engelberth wrote: In order to reconnect to asterisk (asterisk -r), you need root permissions. In order to use 'asterisk -r' you need write access to the unix-domain socket /var/run/asterisk/asterisk.ctl . Root has it. Others may have it as well. The error message above has nothing to do with it, however. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Invite + decreasing sequence number = 500 Error?
Hi out there We have a strange Problem here with invites. We observe this SIP conversation. C3 PBX - Asterisk Case 1. Sequence Numer always increasing: = Invite = 401 Unauthenticated = Invite+auth with sequence number previous Invite. = 100 Trying etc. Works OK. Case 2. Sequence Number decreasing. = Invite = 401 Unauthenticated = Invite+auth with sequence number previous Invite. = 500 ERROR I was browsing the SIP rfc and I cannot find any clue if in this case the sequence numbers must be increasing (the C3 PBX is wrong) or if I have sumbled over an asterisk bug. Is there anyone who knows? Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch __ -- _ -- 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] Far end nat traversal for media is not working always
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 We use a obfuscation software to encrypt/mangle both SIP/RTP which sits before asterisk. What happens is sometimes we don't get any voice. after some rtp set debug we found out that when received ip of the rtp stream is router's public ip, everything works cleanly. But sometimes we get the private ip's of the client as received address in rtp stream which results in no voice. it seems asterisk because of some unknown reason failed to traverse nat for the media stream. What reason behind this strange behavior is still unknown to us. Thanks in advance. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.11 (GNU/Linux) iQEcBAEBAgAGBQJPjA6JAAoJEO6M4UDNbNCeOTsIAJMnKM8J1kRqx3Eqcnk2b89U YxVeGSfurCX87qdJdM4xNxndEzVm9BkDq6kApBB3O5lbV0Mrh06kzkrTVuq3CZwh UbL1TmO7iV4wNrvv9Gl+p9F+2R/pCYQUWFCXyQ6hYqh3rWEgIfB2fQ9xQWqiaW0X q6jQA29G3tstnoDnpR3+eNtTvhrIiDQmcLELGj3MmTYrk2+BuDyLPV431tDEg5i1 uzSqvJI3zQLH2x6CFRnTGE+XPw3zLsBCDatD0LXWvpavXicOthRbX+qREO8M7xW5 y9WP9NkrqRE7hfshbB1VKvNGXj6kmtLpze0WOenZOmCaXkHFdWPXxGCXwvgZlUc= =8aC5 -END PGP SIGNATURE- -- _ -- 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] Caller ID problem
Hi Arthur, read the article and understand,thanks :) Btw, is there any patch for this problem without need to upgrade to version 10.x ? On 4/16/12, Arthur Stanfield a...@dmcip.com wrote: Hi Anam, Hope this helps explain Asterisk version numbering: http://leifmadsen.wordpress.com/2011/08/29/asterisk-10-asterisk-1-hh10/ Easy to get confused!. Cheers, AJ. - Original Message - From: Satria Anamarta anam.satri...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 16 April, 2012 12:10:27 PM Subject: Re: [asterisk-users] Caller ID problem Thanks Danny. I test it with blind transfer and hey, you're right, the caller ID passed successfully, but the attended transfer doesn't. What version did you refer to by saying 10.x ? Asterisk? Shoudn't current version of asterisk is 1.x and should move to 2.x instead of a big jump to 10.x ? Thanks :) BR, Anam Totally newbie On 4/16/12, Danny Nicholas da...@debsinc.com wrote: Do a blind transfer instead of attended transfer - the under the covers changes in 10.X handle this for attended transfers, but to the best of my knowledge, the blind transfer is the only solution in the 1.X tree. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Sunday, April 15, 2012 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Caller ID problem Hi, I'm running asterisk 1.8.7.0 FreePBX 2.8.1 IP Phone Yealink T20 Trustrpid and sendrpid is on the sip.conf Let say I pickup a call on ext A using *8, the caller's caller ID successfully passed to my phone. I decide to pass the call to ext B. On phone B, it display ext A not the original's caller ID. I want on phone B it display the caller's caller ID. Is there any solution for this? I already googling this for around a week but found no solution yet :( Thanks and BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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] When CALL-ID were same , I could hijack another session
Hello all. I want to know this issue is bug or not. My Asterisk version is 1.6.2.6. I used nat=yes on sip.conf. ## Issue 1. SDP session handring by Asterisk ## I used 2 clients , A and B. 2 UAC under another NAT. /// --- router A Asterisk --- router B /// All IP address are examples. Asterisk 155.0.0.* A 192.168.0.2 via 134.255.1.* B 192.168.0.2 via 135.223.10.* Asterisk and A and B have grobal address. A and B are under NAT,and has local address. URI is not same. A AAA@155.0.0.* B BBB@155.0.0.* CALL-ID is same. Both CALL-ID is KKK@192.168.0.2 . After A and Asterisk 's call was established, New call from B will be at last failed. But when I saw B and Asterisk 's SDP log, this was repeated. === INVITE from B to Asterisk Trying Ringing 200 OK from Asterisk to B INVITE from B to Asterisk . . === Call was not began. I think it is true handring at the same CALL-ID. But I can't understand. Why Asterisk returns 200 OK ? Is this correct ? ### Issue 2. On meetme , I can hijack another session. ### I used 4 clients , A and B and C and D. 2 UAC under another NAT. /// C --- router A Asterisk D --- router B /// A and C join in meetme on Asterisk. room 100 B and D join in meetme on Asterisk. room 200 The room was not same. All other setting was same as Issue 1. After A and Asterisk 's call was established, And after C and Asterisk 's call was established, A and C could talk on room 100. Then, new call from B. On Asterisk log, log =full this was repeated. === INVITE from B to Asterisk Trying Ringing 200 OK from Asterisk to B INVITE from B to Asterisk . . === It looked B's call has failed. But It was not failed !! B could hear the voice of A and C conference. Is this collect ?? Why I can hear another room's conference? Is this mean session hijack ?? I could do this. I want to know how to prevent this. any help appreciated. nakaji -- _ -- 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] Caller ID problem
No - if someone figures out a way, let me know since my receptionist doesn't like blind transfers. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Monday, April 16, 2012 7:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID problem Hi Arthur, read the article and understand,thanks :) Btw, is there any patch for this problem without need to upgrade to version 10.x ? On 4/16/12, Arthur Stanfield a...@dmcip.com wrote: Hi Anam, Hope this helps explain Asterisk version numbering: http://leifmadsen.wordpress.com/2011/08/29/asterisk-10-asterisk-1-hh10/ Easy to get confused!. Cheers, AJ. - Original Message - From: Satria Anamarta anam.satri...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 16 April, 2012 12:10:27 PM Subject: Re: [asterisk-users] Caller ID problem Thanks Danny. I test it with blind transfer and hey, you're right, the caller ID passed successfully, but the attended transfer doesn't. What version did you refer to by saying 10.x ? Asterisk? Shoudn't current version of asterisk is 1.x and should move to 2.x instead of a big jump to 10.x ? Thanks :) BR, Anam Totally newbie On 4/16/12, Danny Nicholas da...@debsinc.com wrote: Do a blind transfer instead of attended transfer - the under the covers changes in 10.X handle this for attended transfers, but to the best of my knowledge, the blind transfer is the only solution in the 1.X tree. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Sunday, April 15, 2012 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Caller ID problem Hi, I'm running asterisk 1.8.7.0 FreePBX 2.8.1 IP Phone Yealink T20 Trustrpid and sendrpid is on the sip.conf Let say I pickup a call on ext A using *8, the caller's caller ID successfully passed to my phone. I decide to pass the call to ext B. On phone B, it display ext A not the original's caller ID. I want on phone B it display the caller's caller ID. Is there any solution for this? I already googling this for around a week but found no solution yet :( Thanks and BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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] SNOM phones? Please test this patch (broken hints with notifycid=yes)
Niccolo: I've reopened the issue and placed some comments on the issue requesting more information. In the future, if you need an issue reopened, you can contact a bug marshal in #asterisk-bugs. Thanks, Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org - Original Message - From: Niccolò Belli darkba...@linuxsystems.it To: asterisk-users@lists.digium.com Sent: Monday, April 16, 2012 3:56:09 AM Subject: [asterisk-users] SNOM phones? Please test this patch (broken hints with notifycid=yes) Patch: https://issues.asterisk.org/jira/secure/attachment/42605/local_remote_hint2.diff https://issues.asterisk.org/jira/browse/ASTERISK-16735?focusedCommentId=191720page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-191720 It's already merged in asterisk 10.4-rc1, it breaks hints for me but I suspect it may be a snom's bug. Cheers, Niccolò -- _ -- 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] [SOLVED] 404 Response to Invite - Should be 401
Hi all, This post is in case someone else has this problem. The cause of the issue turned out to be one of the site technicians having the same extension registering from his laptop as the ATA we were testing. His laptop wasn't always connected to the voice network and the soft-phone wasn't always on. Sometimes we were able to make calls from the ATA we were testing and sometimes we would get the problem described below. Everything came to light when his soft-phone registered (throwing up a non-voice network IP address) whilst I was watching the CLI and we realised the problem... Hope this helps someone. -- Forwarded message -- Date: 1 April 2012 23:42 Subject: 404 Response to Invite - Should be 401 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi all, I am currently testing a new version of firmware released by an ATA vendor and I have come across a strange problem in 1.8.9.0. Sometimes if I dial immediately after hanging up a previous call (we used voicemail for our testing as it has unlimited capacity) Asterisk will return a 404 code instead of doing the usual INVITE - 401 - INVITE sequence. The CLI says that the call failed because the extension was not found in context 'default'. It appears from this (as the ATA is correctly setup and normally make calls to a different context) that Asterisk believes the extension is unauthenticated for some calls and sending them straight to the guest (default) context. What should I be looking for in the initial invite from the ATA which will be triggering a 404? Is there something wrong in the SIP header / body which would be triggering a safety lock down of the call? I have tried to look at the invite packets but I can't see anything that is incorrect. On one occasion the sequence was INVITE - 401 - INVITE - 404 but most commonly it is INVITE - 404. We have also noticed that sometimes the ATA stops responding to OPTIONS requests (qualify) whilst this issue is happening and there may be some other registration issues. Some diagnostics have already been completed but so far we haven't found anything which points us to where the issue actually is. We assume it is an ATA related issue. Any pointers and suggestions greatly appreciated. -- _ -- 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] Invite + decreasing sequence number = 500 Error?
It's not a bug - decrementing the CSeq header field value is directly in violation of RFC 3261. From section 22.2: When a UAC resubmits a request with its credentials after receiving a 401 (Unauthorized) or 407 (Proxy Authentication Required) response, it MUST increment the CSeq header field value as it would normally when sending an updated request. Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org - Original Message - From: Benoit Panizzon benoit.paniz...@imp.ch To: asterisk-users@lists.digium.com Sent: Monday, April 16, 2012 7:12:09 AM Subject: [asterisk-users] Invite + decreasing sequence number = 500 Error? Hi out there We have a strange Problem here with invites. We observe this SIP conversation. C3 PBX - Asterisk Case 1. Sequence Numer always increasing: = Invite = 401 Unauthenticated = Invite+auth with sequence number previous Invite. = 100 Trying etc. Works OK. Case 2. Sequence Number decreasing. = Invite = 401 Unauthenticated = Invite+auth with sequence number previous Invite. = 500 ERROR I was browsing the SIP rfc and I cannot find any clue if in this case the sequence numbers must be increasing (the C3 PBX is wrong) or if I have sumbled over an asterisk bug. Is there anyone who knows? Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch __ -- _ -- 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] Custom Application recording problem
Greetings All,
I have a compatibilty problem between asterisk 1.4 and 1.6.2
In my 1.4 asterisk I have a custom application that users call and make
recordings which recording I save to a file with the caller Id.
Below is the config file which works perfectly in 1.4
[timo]
exten = 3552,1,Set(TIMEOUT(digit)=2) ; Set Digit Timeout to 5
seconds
exten = 3552,2,Set(TIMEOUT(response)=2) ; Set Response Timeout to
10 seconds
exten = 3552,3,Answer
exten = 3552,4,NoOp(${CALLERID(num)})
exten = 3552,5,Set(number=${CALLERID(num)})
exten = 3552,6,NoOp(${number})
exten = 3552,7,Background(recmsg1) ;Please say yo message after the beep
and end with a hash
exten = 3552,8,Record(crystalrecords/${number}.gsm)
exten = 3552,9,Playback(crystalrecords/${number})
exten = 3552,10,Background(ackrec) ;Press 1 to replay or 2 to
re-record, 3 to save
exten = 3552,11,WaitExten(5)
exten = timo,1,1,Goto,timo|3552|9
exten = timo,2,1,Goto(3552,7) ; re-record message
exten = timo,3,1,Goto(4,1)
exten = timo,4,AGI(timorec.php)
exten = i,1,Background(invalidentry)
exten = i,n,Goto(3552,10)
exten = t,1,Playback(thankyoubye)
exten = t,n,Hangup
In my 1.6 version I use the same configuration in extensions_custom.conf but
I get the error below. It seems like 1.6 does not recognize the button the
user has pressed.
The specific error is
-- Invalid extension '1' in context 'from-internal' on SIP/440-004b
The detailed log is below.
-- Executing [3552@from-internal:1] Set(SIP/440-004b,
TIMEOUT(digit)=2) in new stack
-- Digit timeout set to 2.000
-- Executing [3552@from-internal:2] Set(SIP/440-004b,
TIMEOUT(response)=2) in new stack
-- Response timeout set to 2.000
-- Executing [3552@from-internal:3] Answer(SIP/440-004b, ) in
new stack
-- Executing [3552@from-internal:4] NoOp(SIP/440-004b, 440) in
new stack
-- Executing [3552@from-internal:5] Set(SIP/440-004b,
number=440) in new stack
-- Executing [3552@from-internal:6] NoOp(SIP/440-004b, 440) in
new stack
-- Executing [3552@from-internal:7] BackGround(SIP/440-004b,
recmsg1) in new stack
-- SIP/440-004b Playing 'recmsg1.gsm' (language 'en')
-- Channel 0/2, span 4 got hangup request, cause 16
== Spawn extension (ivr-16, s, 12) exited non-zero on 'DAHDI/95-1'
-- Executing [h@ivr-16:1] Hangup(DAHDI/95-1, ) in new stack
== Spawn extension (ivr-16, h, 1) exited non-zero on 'DAHDI/95-1'
-- Hungup 'DAHDI/95-1'
-- Executing [3552@from-internal:8] Record(SIP/440-004b,
crystalrecords/440.gsm) in new stack
-- SIP/440-004b Playing 'beep.gsm' (language 'en')
-- Executing [3552@from-internal:9] Playback(SIP/440-004b,
crystalrecords/440) in new stack
-- SIP/440-004b Playing 'crystalrecords/440.gsm' (language 'en')
-- Executing [3552@from-internal:10] BackGround(SIP/440-004b,
ackrec) in new stack
-- SIP/440-004b Playing 'ackrec.gsm' (language 'en')
-- Invalid extension '1' in context 'from-internal' on SIP/440-004b
== CDR updated on SIP/440-004b
-- Executing [i@from-internal:1] BackGround(SIP/440-004b,
invalidentry) in new stack
-- SIP/440-004b Playing 'invalidentry.slin' (language 'en')
== Spawn extension (from-internal, i, 1) exited non-zero on
'SIP/440-004b'
-- Executing [h@from-internal:1] Macro(SIP/440-004b, hangupcall)
in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(SIP/440-004b,
1?noautomon) in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp(SIP/440-004b,
TOUCH_MONITOR_OUTPUT=) in new stack
-- Executing [s@macro-hangupcall:4] GotoIf(SIP/440-004b,
1?noautomon2) in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] NoOp(SIP/440-004b,
MONITOR_FILENAME=) in new stack
-- Executing [s@macro-hangupcall:7] GotoIf(SIP/440-004b,
1?skiprg) in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf(SIP/440-004b,
1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,13)
-- Executing [s@macro-hangupcall:13] GotoIf(SIP/440-004b,
1?theend) in new stack
-- Goto (macro-hangupcall,s,15)
-- Executing [s@macro-hangupcall:15] Hangup(SIP/440-004b, ) in
new stack
== Spawn extension (macro-hangupcall, s, 15) exited non-zero on
'SIP/440-004b' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/440-004b'
-- Remote UNIX connection
-- Remote UNIX connection disconnected
Kind Regards
Billy
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[asterisk-users] Dial Local doesn't honore the channel language setting
Hi,
on asterisk 1.6.2.22
exten = s,1,Set(CHANNEL(language)=fr)
exten = s,n,Dial(Local/${myEXTEN}@context/n,,)
Transfer, Voicemail, demo, aso are played in english even despite the
fact that language=fr in sip.conf.
What's wrong? Bug?
Thanks for any hint
--
Daniel
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[asterisk-users] 10.3 : sip loses registration ?
We found this morning we had no SIP connection to another site. sip show registry on the main site gave no authentication. sip show peers on the other site showed the peer unspecified. The odd part about this: doing sip reload on the main site made it all work. Nothing else was changed. Main site: sip show registry SFO:5060 N sip_outgoin 105 No AuthenticationSat, 14 Apr 2012 14:48:15 4 SIP registrations. .. PBX*CLI sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found == Using SIP TOS bits 96 == Using SIP CoS mark 3 == Parsing '/etc/asterisk/sip_notify.conf': == Found == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 On other site: sip show peers . sip_outgoing/s (Unspecified)D N 0Unmonitored .. 13 sip peers [Monitored: 3 online, 7 offline Unmonitored: 2 online, 1 offline] result of sip reload on main site: -- Registered SIP 'sip_outgoing' at xxx.yyy.zzz.aaa:9345 How do I/can I set the main site to retry a registration? I've now changed sip.conf to add: registertimeout=20 registerattempts=0;Default is 0 tries, continue forever But these are the defaults anyhow! Thanks, sean -- _ -- 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 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
On 04/14/2012 07:33 AM, Niccolò Belli wrote: Il 04/04/2012 07:45, Anton Kvashenkin ha scritto: Check it out, thank you. You're welcome. New packages against dahdi-linux-2.6.0, dahdi-tools-2.6.0, libpri 1.4.12+svn20120409 and spandsp-0.0.6~pre20: http://www.linuxsystems.it/2012/04/asterisk-1-8-11-0-debian-squeeze-packages-with-t-38-gateway-queue-hints-and-fixed-rfc4235/ If someone can help me there is a bug with T38 gw and eutelia: http://lists.digium.com/pipermail/asterisk-dev/2012-April/054681.html Asterisk 10.4-rc1 does work, so it should be a matter of identifying the problem and backporting the fix. Keep in mind that the T.38 gateway code was reworked rather substantially when it was merged into Asterisk 10; the last version that irroot published for Asterisk 1.8 was long before this rework occurred. It's quite unlikely that locating a simple difference will actually occur, or that it would be easy to backport. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
Re: [asterisk-users] 10.3 : sip loses registration ?
I have experienced this issue with a provider with Asterisk 1.2, 1.6 1.8. I never got to the root cause of the problem however it used to occur quite frequently, now it appear to occur once every month or two - haven't seen it occur for a while now but then I have been incrementally updating my version of asterisk, currently 1.8.11.0 The preceding events I observed was that there would be a timeout communicating with the peer followed by retry attempts and finally a message reporting Wrong password, this is the point at which registration attempts stopped despite the value in sip.conf being set to 0. As per your observations a 'sip reload' gets things going again. When the problem was occurring within a 24-hour period I set up an SPA-942 phone to register to the service and captured packets between them, I don't recall seeing any issues over a period of a few days with the SPA phone hence was baffled by this phenomenon and have been since. I was considering writing a script to check for the No Authentications status and to then issue a 'sip reload' but as the problem is rarely seen now I haven't had to do this. My suspicion to the cause of the problem is that the authentication database at the VSP may have been offline momentarily hence why the response of a wrong password, I wasn't convinced of this as the packet capture of the SPA-942 did not reveal any authentication errors. Cheers, Larry. On 16/04/2012 10:26 PM, sean darcy wrote: We found this morning we had no SIP connection to another site. sip show registry on the main site gave no authentication. sip show peers on the other site showed the peer unspecified. The odd part about this: doing sip reload on the main site made it all work. Nothing else was changed. Main site: sip show registry SFO:5060 N sip_outgoin 105 No AuthenticationSat, 14 Apr 2012 14:48:15 4 SIP registrations. .. PBX*CLI sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found == Using SIP TOS bits 96 == Using SIP CoS mark 3 == Parsing '/etc/asterisk/sip_notify.conf': == Found == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 On other site: sip show peers . sip_outgoing/s (Unspecified)D N 0Unmonitored .. 13 sip peers [Monitored: 3 online, 7 offline Unmonitored: 2 online, 1 offline] result of sip reload on main site: -- Registered SIP 'sip_outgoing' at xxx.yyy.zzz.aaa:9345 How do I/can I set the main site to retry a registration? I've now changed sip.conf to add: registertimeout=20 registerattempts=0;Default is 0 tries, continue forever But these are the defaults anyhow! Thanks, sean -- _ -- 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] Custom Application recording problem
On 04/16/2012 08:36 AM, Billy Kaye wrote: In my 1.4 asterisk I have a custom application that users call and make recordings which recording I save to a file with the caller Id. Below is the config file which works perfectly in 1.4 I am not going to say that your application doesn't work under 1.4, but to me it looks like it shouldn't work under 1.4. The issue is that you do not have an extension '1' defined within your context of [timo]. (Not to mention your CLI output appears to be from a different context all together.) When the user presses 1, Asterisk cannot find a valid extension to send the caller to. The reason is these lines are not valid. exten = timo,1,1,Goto,timo|3552|9 exten = timo,2,1,Goto(3552,7) ; re-record message exten = timo,3,1,Goto(4,1) exten = timo,4,AGI(timorec.php) If Asterisk even parses them at all, they would define an extension 'timo' with 4 priorities. I suspect they should be... exten = 1,1,Goto,timo|3552|9 exten = 2,1,Goto(3552,7) ; re-record message exten = 3,1,Goto(4,1) exten = 4,AGI(timorec.php) Dale -- _ -- 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 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
I applied the patch to my 1.8.11.0 build and observed the same error as shown in you t38_send.log. I have maintained a private patch file for this functionality and reverted to it when I too observed the INTERNAL_OBJ: user_data is NULL message. Do you have directmedia=no in your SIP configuration? Cheers, Larry. On 14/04/2012 8:33 PM, Niccolò Belli wrote: Il 04/04/2012 07:45, Anton Kvashenkin ha scritto: Check it out, thank you. You're welcome. New packages against dahdi-linux-2.6.0, dahdi-tools-2.6.0, libpri 1.4.12+svn20120409 and spandsp-0.0.6~pre20: http://www.linuxsystems.it/2012/04/asterisk-1-8-11-0-debian-squeeze-packages-with-t-38-gateway-queue-hints-and-fixed-rfc4235/ If someone can help me there is a bug with T38 gw and eutelia: http://lists.digium.com/pipermail/asterisk-dev/2012-April/054681.html Asterisk 10.4-rc1 does work, so it should be a matter of identifying the problem and backporting the fix. Thanks, Niccolò -- _ -- 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] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
Hi, Il 16/04/2012 22:50, Larry Moore ha scritto: Do you have directmedia=no in your SIP configuration? Yes I have. Niccolò -- _ -- 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] Custom Application recording problem
Thanks Dale, Am not sure why it was working in 1.4 but for some reason it was ( Note : My Asterisk is running bundled with Elastix). But any your suggestion worked very fine. Now am having one problem how can define those extensions only with in different contexts, the problem I see is since am Building 3 recording applications only one will be able call its AGI file, Say if someone calls custom extension 1114 They can record message -Press 1 to Replay Press 2 to Re-record or Press 3 to Save the file Also if someone calls custom extension 1115 -Press 1 to Replay Press 2 to Re-record or Press 3 to Save the file Note Each save file selection calls a different AGI file E.g exten = 1,1,Goto,timo|3552|9 exten = 2,1,Goto(3552,7) ; re-record message exten = 3,1,Goto(4,1) exten = 4,AGI(timorec.php) Kind Regards Billy On 4/16/12 11:22 PM, Dale Noll dn...@wi.rr.com wrote: On 04/16/2012 08:36 AM, Billy Kaye wrote: In my 1.4 asterisk I have a custom application that users call and make recordings which recording I save to a file with the caller Id. Below is the config file which works perfectly in 1.4 I am not going to say that your application doesn't work under 1.4, but to me it looks like it shouldn't work under 1.4. The issue is that you do not have an extension '1' defined within your context of [timo]. (Not to mention your CLI output appears to be from a different context all together.) When the user presses 1, Asterisk cannot find a valid extension to send the caller to. The reason is these lines are not valid. exten = timo,1,1,Goto,timo|3552|9 exten = timo,2,1,Goto(3552,7) ; re-record message exten = timo,3,1,Goto(4,1) exten = timo,4,AGI(timorec.php) If Asterisk even parses them at all, they would define an extension 'timo' with 4 priorities. I suspect they should be... exten = 1,1,Goto,timo|3552|9 exten = 2,1,Goto(3552,7) ; re-record message exten = 3,1,Goto(4,1) exten = 4,AGI(timorec.php) Dale -- _ -- 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] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
Perhaps your problem may be that Asterisk doesn't like to send T.38 to a peer other than the one it negotiates the SIP connection with. If I recall correctly you mentioned a while back that eutelia made a change which broke your outgoing T.38 functionality, did you ever find out what the change was? Larry. On 17/04/2012 4:58 AM, Niccolò Belli wrote: Hi, Il 16/04/2012 22:50, Larry Moore ha scritto: Do you have directmedia=no in your SIP configuration? Yes I have. Niccolò -- _ -- 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
