[asterisk-users] SIP transfer issue
Wondering if anyone on here can help with a niggling issue: One of our extensions is unable to make attended transfers at all. The phone in question is an Elmeg ip290, and works fine for direct transfers. However, on attempting to make an attended transfer, the first leg succeeds (the inbound call is placed on hold and gets MoH, the Elmeg user announces the call to the target extension), but upon completing the transfer, both parties get MoH, not each other. There is an entry in the asterisk logs as follows: chan_sip.c:6930 get_refer_info: Supervised transfer requested, but unable to find callid '[EMAIL PROTECTED]'. Both legs must reside on Asterisk box to transfer at this time. The incoming call, the Elmeg and the target extension are all on the same asterisk box. The Elmeg is behind NAT, but canreinvite=no and nat=yes are both set in the appropriate sip.conf sections for both the Elmeg and the target destination. Can anyone shed any light on this? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP transfer from agent fails
I have seen a couple of posts related to this, but no workaround. Setup; Asterisk 1.2.13 with Polycom IP501 phones Caller is sent to the queue with the t option Agent is logged in via AgentCallbackLogin on an extension that is in a context that includes exclusively agent extensions. Agent is set up with ackcall=yes (# to answer) Call comes in, agent takes the call, attempts to transfer to another extension using a SIP transfer on the Polycom phone. Call drops when completing the transfer. The caller goes on hold as they should, the second call is dialed and answers successfully, but the completion of the transfer fails and the call is dropped. There is a internal server error 500 logged on the console from the phone of the agent the originally answered the call. This used to work in earlier versions, quit working some time around 1.2.10 I think. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip transfer, Sip on hold
Hi everybody, sorry for my english but i'm italian and i don't know it very well. I'm trying to do a java-program to traduce and notify asterisk events to a Tapi program. I've a problem with call trasfer. When i transfer a sip user i would like to put his line on hold but i can't do it. He listen the music on hold but his state is connected and not Hold. I hope you understand my problem :-) Thank you very much Bye, K ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip transfer, Sip on hold
9 jun 2006 kl. 10.18 skrev Nicola Pascelupo: Hi everybody, sorry for my english but i'm italian and i don't know it very well. I'm trying to do a java-program to traduce and notify asterisk events to a Tapi program. I've a problem with call trasfer. When i transfer a sip user i would like to put his line on hold but i can't do it. He listen the music on hold but his state is connected and not Hold. At this point, the SIP channel and Asterisk does never put a phone on hold, we play music on hold music. We discussed this at a recent developer meeting and are looking to implement a way to set an option per device whether you want to play music or actually signal hold status to the other end. Right now, the other end will never know that it's on hold, it just gets another audio stream and merrily continues the call. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP transfer/REFER to voicemail problem
For anyone else who might run into this, I got around the transferring to voicemail problem by putting a canreinvite=no line into the section for each caller's SIP address in sip.conf. Not ideal, but it works. I also had to add a dtmfmode=inband for my Mediatrix 1204 addresses to be able to access the voicemail commands. -- I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't answer. Caller (A) hears the Asterisk voicemail prompts, but the voicemail application doesn't hear any audio or DTMF. Easy to duplicate: 1.) A - B (INVITE) 2.) B - C (REFER A to C) 3.) A - C More descriptive: 1.) Caller (A) calls intermediary (B). (B can be any SIP user agent) 2.) Intermediary (B) REFERs caller (A) to callee (C) 3.) C is either a SIP UA which times-out and Asterisk takes to Voicemail, or an extension tied to VoicemailMain. I've come across a thread saying that the Asterisk voicemail system only uses the GSM codec, but if this were the problem, then how can the caller (using mu-law) hear the voicemail prompts? Would Asterisk be doing a half duplex protocol conversion? Any insight would be greatly appreciated!! Current configuration: Fedora Core 1 Asterisk - 1.0.7 (had same problem on 1.0.6) SJPhone - 1.50.271d, Mar 11 2005 (WinXP) XLite - 1103m build stamp 14262 (WinXP) Zultys Zip2 - ZUTS 3.52 sip.conf exerpt: [6003] ; (A) type=friend regexten=6003 username=6003 host=dynamic disallow=all ;allow=gsm allow=ulaw [6004] ; (C) type=friend regexten=6004 username=6004 host=dynamic disallow=all ;allow=gsm allow=ulaw [2101] ; (B) type=friend regexten=2101 username=2101 host=dynamic disallow=all ;allow=gsm allow=ulaw extensions.conf exerpt: exten = 6003,1,Dial(SIP/1003,15) exten = 6003,2,Voicemail(u1003) exten = 6003,102,Voicemail(b1003) exten = 6004,1,Dial(SIP/1004,5) exten = 6004,2,Voicemail(u1004) exten = 6004,102,Voicemail(b1004) exten = 2101,1,Dial(SIP/2101) exten = 8500,1,VoicemailMain exten = 8500,2,Hangup Asterisk (-dvvgc) with sip debug on (REFER-ing caller to VoicemailMain) : -- No username but # key pressed. Using CID '6003' -- Playing 'vm-password' (language 'en') Urgent handler -- Incorrect password '' for user '6003' (context = ,any) -- Playing 'vm-incorrect-mailbox' (language 'en') Urgent handler __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP transfer/REFER to voicemail problem
I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't answer. Caller (A) hears the Asterisk voicemail prompts, but the voicemail application doesn't hear any audio or DTMF. Easy to duplicate: 1.) A - B (INVITE) 2.) B - C (REFER A to C) 3.) A - C More descriptive: 1.) Caller (A) calls intermediary (B). (B can be any SIP user agent) 2.) Intermediary (B) REFERs caller (A) to callee (C) 3.) C is either a SIP UA which times-out and Asterisk takes to Voicemail, or an extension tied to VoicemailMain. I've come across a thread saying that the Asterisk voicemail system only uses the GSM codec, but if this were the problem, then how can the caller (using mu-law) hear the voicemail prompts? Would Asterisk be doing a half duplex protocol conversion? Any insight would be greatly appreciated!! Current configuration: Fedora Core 1 Asterisk - 1.0.7 (had same problem on 1.0.6) SJPhone - 1.50.271d, Mar 11 2005 (WinXP) XLite - 1103m build stamp 14262 (WinXP) Zultys Zip2 - ZUTS 3.52 sip.conf exerpt: [6003] ; (A) type=friend regexten=6003 username=6003 host=dynamic disallow=all ;allow=gsm allow=ulaw [6004] ; (C) type=friend regexten=6004 username=6004 host=dynamic disallow=all ;allow=gsm allow=ulaw [2101] ; (B) type=friend regexten=2101 username=2101 host=dynamic disallow=all ;allow=gsm allow=ulaw extensions.conf exerpt: exten = 6003,1,Dial(SIP/1003,15) exten = 6003,2,Voicemail(u1003) exten = 6003,102,Voicemail(b1003) exten = 6004,1,Dial(SIP/1004,5) exten = 6004,2,Voicemail(u1004) exten = 6004,102,Voicemail(b1004) exten = 2101,1,Dial(SIP/2101) exten = 8500,1,VoicemailMain exten = 8500,2,Hangup Asterisk (-dvvgc) with sip debug on (REFER-ing caller to VoicemailMain) : -- No username but # key pressed. Using CID '6003' -- Playing 'vm-password' (language 'en') Urgent handler -- Incorrect password '' for user '6003' (context = ,any) -- Playing 'vm-incorrect-mailbox' (language 'en') Urgent handler __ Discover Yahoo! Find restaurants, movies, travel and more fun for the weekend. Check it out! http://discover.yahoo.com/weekend.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip transfer and redirect in a Company setting
If I understand your problem correctly, you have user a setup with vm
box a, and user b with vm box b, when secretary uses local callFWD
from phone a to phone b, vm of b picks up. And you want that if it was
redirected from phone a vm box of a should answer. I think (I never
tested this) that the RDNIS variable (${RDNIS}) will hold the CallerID
of phone a, which you can use in your dialplan to use for voicemail if
it exists, something like this will do:
exten = _1XX,1,Dial(SIP/${EXTEN},45,tr)
exten = _1XX,2,GotoIf($[${RDNIS} 0 ]?10)
exten = _1XX,3,VoiceMail(u${EXTEN})
exten = _1XX,10,VoiceMail(u${RDNIS})
I'm not sure if DNID or RDNIS will work for SIP phones, but one of
those should work.
Another way to get this done (ugly), is to set a variable for the
channel before you use the Dial command, like this
exten = _1XX,1,SetVar(ORIGINAL_EXTEN=${EXTEN})
and then test if ${ORIGINAL_EXTEN} is different than ${EXTEN}
Look at this:
http://bugs.digium.com/bug_view_page.php?bug_id=0002590
this:
http://bugs.digium.com/bug_view_page.php?bug_id=0002763
and this:
http://www.voip-info.org/wiki-RDNIS
I hope this helps.
On 4/11/05, Jeb Campbell [EMAIL PROTECTED] wrote:
I have an asterisk box setup and dialplan that is something like this:
(t1/pri)
|
[incoming]
1234,1,Dial(SIP/secretary,30,rt)
1234,2,Voicemail([EMAIL PROTECTED])
Now the t in the dial lets the sec transfer with # and if the person
transferred to is unavail it goes to their voicemail -- that works great.
However if the sec tells her phone to redirect to another phone (CFWDall
on a 7960) asterisk will redirect that call to that phone. However it
uses the sec's context to dial, which if redirecting internally included
voicemail.
So if the sec redirects to another phone and that phone does not answer,
the redirected phone's voicemail plays and not the companies.
I just wanted to see if anyone else had this problem (and a solution).
Jeb Campbell
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip transfer and redirect in a Company setting
I have an asterisk box setup and dialplan that is something like this: (t1/pri) | [incoming] 1234,1,Dial(SIP/secretary,30,rt) 1234,2,Voicemail([EMAIL PROTECTED]) Now the t in the dial lets the sec transfer with # and if the person transferred to is unavail it goes to their voicemail -- that works great. However if the sec tells her phone to redirect to another phone (CFWDall on a 7960) asterisk will redirect that call to that phone. However it uses the sec's context to dial, which if redirecting internally included voicemail. So if the sec redirects to another phone and that phone does not answer, the redirected phone's voicemail plays and not the companies. I just wanted to see if anyone else had this problem (and a solution). Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip transfer
Good day all Is it possible to transfer sip calls?And how? I saw transfer in iax.conf? Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Transfer problem
It's strange to reply to my own email. So please see below of new problem with transfers. - Original Message - From: Ariel's M-tech account To: [EMAIL PROTECTED] Sent: Friday, January 30, 2004 11:55 AM Subject: [Asterisk-Users] SIP Transfer problem I have been following and reading about the SIP problem of transferring calls with Asterisk. I did not see this problem as having a fix or having a patch for it. I can not use the # in our system due to IVR systems we access. I have found that transfer to an extension other then parking works just fine. What is broken is trying to park the call. On sip phones I am able to transfer to meetme, voicemail and other sip or zap ports. But not to the parking extension. So does someone know how to get this working? Can someone let me know at what stage this is at. This is a major problem with our system in deploying SIP phones. We have Cisco 7960, Snom 200 and IpDialog's working but can not transfer. Thank you ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Transfer problem
I have been following and reading about the SIP problem of transferring calls with Asterisk. I did not seethis problem as havinga fix orhaving apatch for it. I can not use the # in our system due to IVR systems we access. Can someone let me know at what stage this is at. This is a major problem with our system in deploying SIP phones. We have Cisco 7960, Snom 200 and IpDialog's working but can not transfer. Thank you
Re: [Asterisk-Users] SIP Transfer
Blind and assisted transfer work with Cisco 7960 phones. Blind transfer works fine with Budgetones. As long as you register to Asterisk. Jamie Carl wrote: Ok, just been thinking about this and thought I would ask before trying it out again. What is the state of SIP transfers? By this I mean transfers initiated via SIP messages, not via DTMF and '#'. Last time I tried, on X-Lite, clicking the transfer button dropped the call. Also, are/will both REFER and BYE/also methods be supported? To me, the SIP way of transfering is alot nicer and it seems silly to me to have a transfer button on your SIP phone that u can't use. Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Transfer
Ok, just been thinking about this and thought I would ask before trying it out again. What is the state of SIP transfers? By this I mean transfers initiated via SIP messages, not via DTMF and '#'. Last time I tried, on X-Lite, clicking the transfer button dropped the call. Also, are/will both REFER and BYE/also methods be supported? To me, the SIP way of transfering is alot nicer and it seems silly to me to have a transfer button on your SIP phone that u can't use. Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Transfer
cvs update -r 1.x channels/chan_sip.c make install where 'x' is from 1 to 30 version 1.30 is dated 2003-04-02 if not sure check rcs2log -v |more regards Martin On Tue, 1 Apr 2003, Russ Beaupre, P.E. wrote: A while ago SIP transfer via the # key on a call to a cell phone via iconnect was working. I updated to the current CVS tonight and now that functionality is gone. Any ideas as to how to enable it again? Thanks in advance -russ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
