similar to: Audio when a call is on hold.

Hello all, I have a problem where problem with one way audio, and I think it's related to "a=sendonly" and a re-invite. Can anyone please assist? The scenario is as follows.... - We send an INVITE to a peer, and it replies with a "100 Trying", and then a "183 Session Progress" message containing "a=sendonly". - Asterisk plays the caller music on hold,

Hi, Recently we got a new feature request from our customer, they want a report to list the duration that agents putting customer on hold, they want to base on this to measure the agents performance. I cannot find any events in cdr, message logs, or manager interface, only when I enable sip debug, then I can see the ReInvite Event in the cli , some thing like the attached logs, is there any

Hello, I've been racking my brain over this for much of the day so I thought the list would probably be more helpful. A few days ago I upgraded from Asterisk 1.2 to Asterisk 1.4.5. Everything appeared to be working properly. However, on the first business day, we realized that when transferring calls (not using call parking, using the built in transfer buttons on a Cisco 7960) would not

Hi All, I have a problem with asterisk and call hold. In the re-invite package when I take the call to the hold, the SDP value "a=sendrecv" is present, according to the rfc3264 the sdp value a must be mark with "sendonly". I've already tried with Asterisk 1.8 and Asterisk 11, but there is the same problem. I've already read all the information about canreinvite and

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 all If Asterisk is bridging a call between two SIP peers and one peer puts the other on hold by means of a re-INVITE with SDP containing a=sendonly, Asterisk will play locally generated MOH instead of relaying the media streamed by the SIP peer which took the hold action. Any ideas how to change that? (This is understandable if the peer is a handset but can be a problem if it is a PBX with

facing problem with originating webrtc calls 1-when iam doing call from webrtc iget ice working <--- SIP read from WS:91.196.158.205:1466 ---> INVITE sip:0669197533 at 77.91.132.9 SIP/2.0 Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315 Max-Forwards: 69 To: <sip:0669197533 at 77.91.132.9> From: "Anton" <sip:1065 at 77.91.132.9>;tag=5i21qaop43 Call-ID:

Hi all, I am configuring asterisk in a cable modem network, using a motorola TM401A. I can make calls from the MTA but I can receive, display the following error: -- Executing [1500 at alberti:1] Dial("OSS/dsp", "MGCP/aaln/1 at 0-13-11-82-bd-a.ssw.intercal.net|30") in new stack [Sep 2 00:10:53] NOTICE[28062]: chan_mgcp.c:3572 mgcp_request: Asked to get a channel of

I am having issues with transfers (SIP/REFER) using Asterisk 1.6. You will find the SIP debug below. There are three phones in this setup. 5253 and 5258 are Aastra 53i telephones, 101 is a standard phone connected through an Audiocodes gateway. All phones are registered in context "phones" and are set to not allow reinvites. All phones can dial each other directly. The dialplan

A previous poster mentioned the same thing, with no response: http://lists.digium.com/pipermail/asterisk-users/2004- December/080161.html Fresh asterisk 1.0.5 install on FC3, started with "make samples", nothing fancy. It's so bland, I'm surprised the list isn't full of people having the same trouble. I have several Uniden UIP200 phones and a single Grandstream BudgetTone

I have a Cisco ATA-188 with two POTS phones and latest stable cvs. In any situations with call waiting (existing connection and calling again) the second call cause both calls to drop. This is the same for "internal" extensions and from external (ZAP and SIP). It seems to be a "502 - The transaction could not be executed, because the endpoint does not have sufficient

I am debugging an intermittent issue of missing audio on calls that come from a SIP provider into our asterisk-11.10 installation. Sometimes, incoming calls from this provider work correctly, with audio streaming in both directions. Other times, with the same setup, the calling party is unable to hear the IVR recording from the asterisk installation, although in fact the streaming is supposed to

-- Luca Pradovera luca.pradovera at gmail.com Hello, sorry, I managed to lose the reply amidst the traffic. What we have here is our application server APP with leg A in AsyncAGI in an Adhearsion application, which after some magic dials leg B on the office PBX through a configured peer. Leg B then decides that user C knows more about the subject, and initiates a blind transfer to C?s phone

Hello, I am having problems with music on hold on grandstream phones. When I press Hold button on grandstream phone this is the debug of sip. But nothing happens, no music. Is it problem of asterisk or grandstream budget phone? Sip read: INVITE sip:1105@192.168.1.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK7fcd3a44e7721b41 From:

I am using asterisk 1.4.18 with a polycom phone. sip.conf has: [532] type=friend username=532 secret=XXX dtmfmode=RFC2833 host=dynamic context=smvoice-sip callerid=532 qualify=no nat=no disallow=all allow=ulaw allow=alaw allow=gsm canreinvite=no I call into the dialplan and try to play demo-congrats and I hear nothing. Firewall is disabled. Everything is on the 192.168.1.X network for this

Hi, I have some problems transfering call from phone to phone with my Asterisk. When I dial Flash I can hear for half a second the dial tone, but it stops suddenly. The other phone hear the on hold music and pressing flash key another time I get back to the previous channel. On the asterisk consolle seems to be all ok, this is whant I can read: asterisk1*CLI> -- Swapping 0 for 1 on

Hello all, I am running * 1.2.6 I have 2 linksys PAP2 with two phones each. Until recently all was good. on Friday I was running 1.2.5 when I added the fourth phone. I have to admit to initially wiring the rj11(crossed wires) wrong the first time but other than that nothing I can think of. Added the appropriate entries in sip.con and on the PAP2. I then tried to call from one line to the

We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call

hi, its possibe to "dont start" music on hold when caller (from sip operator trunk) press HOLD (i.e. on mobile phone) Asterisk acts on SDP a=sendonly i want pass trough media from SIP trunk provider Marek

Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the