search for: unhold

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 SIP/2.0 / 102 INVITE 11. CancelDestroy 12. Unhold SIP/2.0 13. TxReq ACK / 102 ACK 14. TxReqRel INVITE / 103 INVITE 15. Rx SIP/2.0 / 103 INV...

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 happens the same with both of them. I've tried several codecs to no luck, canreinvite=no, but nothing does the trick. This is my sip.conf: [general] context= bog...

Hi, I want to write API for doing some actions. I want to write function for hold special call via AMI.But I can not find any action for this purpose. Is there any action for holding special channel? Regards, Mahdieh Saeed -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 sends 200 OK with SDP indicating...

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 exceeded on call 0457bd4153a66af6241b2ce5217e823@192.168.0.3 for seq...

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

...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 cant' seem to find anything on the WIKI about it just sorta locking up like this. BTW if i hold/unhold the extension the audio will come back for a few more seconds..... *CLI> show version Asterisk CVS-05/31/04-22:00:51 built by tretkowski@rollcage on a i686 running Linux *CLI> using ztdummy for timing. -- Undocumented Features quote of the moment... "It's not the one bullet with...

...m 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 happening . When the call is put in hold , i need to update the satus as "CAll is in Hold". For this i need to catch the call hold event . How can i make this ...reply me .... Thanks in advance sathish. ----------------------------...

...first leg second leg What I want to do is putting first call leg on HOLD(My own pre-recorded IVR message) And meanwhile dialing out the second leg.When Asterisk detect "Ringing" on the second leg ( from GSM provider) , the first call leg should be unhold and then Bridging will be occured. Appreciate any help, Mohammad Mirzaee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080518/39f2d21b/attachment.htm

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

...or ignore this signaling on the zap channel(s) ?) Kind regards Thanks NB: Oddly enough, when the local user hears the music on hold, his own channel (a local SIP phone in this case) isn't reported as "On Hold" when issuing "sip show channels" in cli, and no AMI Hold/Unhold events are generated. I double checked, the MOH that gets played is the one specified in sip.conf, NOT zapata.conf.

Thanks Richard. This is exactly the answer I was looking for. I'm now assuming that Asterisk 11 was using it's equivalent "bridge_simple" but I was getting confused because the only bridge module I saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that would have been the only bridge getting loaded at first. Is it expected that if bridge_softmix handled a

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

On 2/23/19 4:19 PM, Joshua C. Colp wrote: > On Sat, Feb 23, 2019, at 11:04 AM, hw wrote: > > <snip> > >> >> directmedia is not explicitly enabled; I guess it's the default. >> >> Joshua basically says there is no way to control which ports are being >> used for SRTP because that it is "up the endpoint". Such endpoints, in >>

On Mon, Jun 29, 2020 at 6:46 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Sun, Jun 28, 2020 at 2:26 PM Dovid Bender <dovid at telecurve.com> wrote: > >> Hi, >> >> We have a box up and we are starting to see a lot of "Exceptionally long >> queue length queuing" in the logs. From all the research so far it seems >> like this leads to

Hi, Does any one experience that SIP phone to SIP phone (Polycom phone) calls can't hear each other, but Monitor application records both end's voices. It also happens in group pickup calls. Zap calls to queue (Local channel) also experience this problem (sometimes, our SIP phone can't hear any voice from incoming Zap calls when pickup, sometimes this happens after 10-50

...have been the only bridge getting loaded at first. > > > Is it expected that if bridge_softmix handled a normal two party call > then MOH would no longer function? > That is correct. bridge_softmix is optimized for multi-party conferencing where passing control frames such as hold/unhold to other parties in the bridge is not a good idea. For example, if three parties are in a bridge and if party A pressed its hold button then that should not necessarily prevent parties B and C from talking to each other. Using bridge_softmix for a normal two party call is a last resort. It works...

...terisk Version : 1.8.9.1 Queue Name : agent-support Agent logged in Extension : SIP/8001,SIP/8002 ... strategy : ringall 1) How to intimate or send information to available agent to inside queue incoming call ringing. 2) How to agent answer incoming call that placed in queue. 3) AMI for agent Hold/Unhold call. Thanks. Rajnikant Vanza -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150417/d29f85eb/attachment.html>

I am using the snom 200 with Phone type snom200-SIP Version snom200-SIP 2.04g Bootloader URL http://www.snom.com/download/snom200-boot1.9.bin Firmware URL http://www.snom.com/download/share/snom200-2.04o-SIP.bin I am using asterisk stable tree. I had to disable "Challenge Response on Phone" on my snom; I could not get it to work with