similar to: Bridging a call on hold with an active call

I got a very strange problem with call-hold function. For calls that come in from PSTN and route to a SIP extension. If I put the call on hold, I cannot unhold the call after. The caller would be left with hold music forever. A warning message would be shown on the console usually a few seconds after putting the call on hold: WARNING[17428]: chan_sip.c:686 retrans_pkt: Maximum retries

I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRel INVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx

i need to catch the call hold event from my asterisk-java program. Im using net.sf.asterisk.*; for communicating with asterisk server. I need to get the call hold status on my java program . I can able to get the music on hold status but i cannot able to get the call hold status. The events like 1. HoldEvent , 2.HoldedcallEvent 3. UnHold event are not getting fired when the call hold is

HI All; I want to use Asterisk for just Voicemail Server and I need a Dynamic creation of Mailboxes. My users 's Mailboxes are same as "Extensions" but I donot want to add mailboxes in "Voicemail.conf" Is there any way to create mailbox from Asterisk dial-plan ? Appreciate any suggestions Mohammad Mirzaee -------------- next part -------------- An HTML attachment was

Hello, I think I met a case similar to the one solved by [1] . Quoting this case : * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk

Hi everybody We have a strange problem with several asterisk servers (Version 1.4.11) using PRI cards (tied to telco here in Belgium). Indeed we noticed that whenever a local user places an outgoing call through the PRI (and telco) to another IPBX (tied to telco using BRI or PRI), if the remote party places the call on hold, the caller hears the _local_ music on hold instead of the

Hello everybody and merry xmas. I have a problem with sip phones calling each other inside the same network (no nat, no firewall). You can make and receive calls and pick them up, but you cannot hear anything on any side of the call. But if you press hold/unhold or you transfer the call, then everything works as expected. Ths SIP phones I've tried are Swissvoice IP10s and kphone, it

Hi, I have the following symptoms: Call X-lite / Nokia E51 X-lite press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH Call X-lite / SCCP phone MOH works as supposed Call SCCP phone / Nokia E51 SCCP press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH In addition, the BLF on the SCCP phones does NOT show the hinted SIP extension on hold. With 1.4

I'm trying to capture when a call is placed on and removed from being on hold through the AMI in Asterisk 12.3. In previous versions, the Hold event contained a 'Status' field which indicated if the call was going 'On' or 'Off' hold. However, in 12 not only am I not seeing the Status field, but I am not seeing any AMI Hold event that corresponds to removing the call

Using X-Lite FWD soft phone, I can register, get to the 'demo' menu extension, but that's it. Audio starts, then after a few seconds stops, with packets still being passed. Anyoen have any clues? Yes there are firewalls between here and there, yes there is NAT at my end...What ports need punching, is rfc2833 the correct settign or should I use inband or info? TIA, I just

Hello, I have been trying to completely disable music on hold on my asterisk system. When a call is put on hold I do not want any music on hold, but I would like the remote user to get informed of this event (depending on the technology e.g. with a SIP reinvite and an SDP indicating the call is on hold). I have searched and tried out various approaches, but when putting the call on hold

HI; Thanks for your reply. The reason for why I am going through asterisk in such case is just "using asterisk voicemail service" I mean: ATA1 calls ATA2, suppose ATA2 is unreachable or he is not at the office, then the call reroute (my GK is able to reroute calls if the first route is not valid) to atersik for voicemail service. Do you think I can handle it with asterisk native

HI All; I want to use Asterisk for just Voicemail Server and I need a Dynamic creation of Mailboxes. My users 's Mailboxes are same as "Extensions" but I donot want to add mailboxes in "Voicemail.conf" Is there any way to create mailbox from Asterisk dial-plan ? Appreciate any suggestions Mohammad Mirzaee Mohammad Mirzaee -------------- next part -------------- An HTML

HI ALL; Anybody can explain the difference between "call parking " vs "call transfer" Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040720/2975991b/attachment.htm

HI ALL; Is there anybody who use app_radius(astersik radius module)??????????? is it stable? Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040803/8a096bfe/attachment.htm

Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X Thanks, jerry

Hello. LCR means least cost routing, and it's billing system problem where to route a call, not b2bua's. But currently I dunno any free billing system that support it, so i moved this logic to b2bua. On Sat, 12 Feb 2005 07:05:39 +0330, mohammad <mohammad@mirzaee.net> wrote: > Hi Mike; > Thanks for your new application, but I think it would be better if you put >

HI ALL; I have problem converting a windows .wav file to .gsm format by Sox. Could anyone help. Cheers, Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/3408bfd5/attachment.htm

HI ALL HI MICHAEL; My name is mohammad and I am iranian.I have been trying to install oh323 channel but I come up with dead end. In fact it makes me crazy. plz help me michael. I saw mailing list and I trid serevel CVS headers such as , 2004-06-07( seven of june) 0r 2004-07-02( second of july) besides I use: 1-openh323 v1.12.2 2-pwlib v1.5.2 3- asterisk CVS (2004-06-07,

Hi, I wonder if it is possible to perform the "find_by_contents" on a subset of fields indexed in acts_as_ferret.If so, how? In my code I have: acts_as_ferret (:fields => [''title'', ''focus'', ''purpose'']) However, I like to have two search options one on all fields and one only on the title. Any help is most appreciated.