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Displaying 20 results from an estimated 80000 matches similar to: "No subject"

2010 Sep 22
1
T38 and codecs negotiation
Hi, I'm working with asterisk 1.4.35 and found an issue regarding codecs negotiation when T38 is enabled (t38pt_udptl=yes). In particular if the INVITE sdp contains no allowed codec the call is not rejected with "488 - Not acceptable here" but it goes through and the 200 OK SDP is as follows: v=0 o=root 27285 27285 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0
2010 Oct 12
0
rtpip patch
Hello *, is the rtpip patch still valid for asterisk 1.6 (with some code changes, obviously)? https://issues.asterisk.org/view.php?id=8161 Or, in asterisk 1.6 there is an alternative to using it? This is the difffile I produced for chan_sip.c in asterisk 1.6.2.11 --- chan_sip.c 2010-10-12 13:47:49.000000000 +0200 +++ chan_sip.c.orig 2010-10-12 13:47:27.000000000 +0200 @@ -987,9 +987,6 @@
2018 May 11
3
SIP Codec negotiation
> On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: >> I receive an INVITE/SDP containing: >> >> m=audio 11310 RTP/AVP 3 0 101 >> >> which I interpret as gsm, ulaw, rfc2833. >> >> and I reply with an OK/SDP containing: >> >> m=audio 15884 RTP/AVP 0 3 101 >> >> which I interpret as ulaw, gsm, rfc2833. >>
2011 Feb 10
2
Unable to make outgoing calls with Internode
Surely there must be someone here who can help me with this problem. I have spent weeks trying to get this damned service to work with no luck. I have incoming calls working, but no outgoing. If get outgoing working then incoming don't work. I have sent this problem to this list a couple of times with little or no response, and I _really_ need some help to sort it out. I have an asterisk
2018 May 11
2
SIP Codec negotiation
On Fri, 11 May 2018, Joshua Colp wrote: >> In the above example, even though the INVITE/SDP says they prefer gsm >> over ulaw and the OK/SDP says I prefer ulaw over gsm, they can choose >> to use gsm or ulaw? > > Yes. > >> Can it be asymmetrical? They send gsm and I send ulaw? > > Technically, yes. In practice it's a bit iffy - specifically because
2020 Jun 12
0
Forbidden call
Hi Steve, - Your right - the file was AMI (copied the other one). By direct connect I simply meant the speaker is an extension on that server. here is the SIP debug <--- SIP read from UDP:X.X.X.X:1024 ---> == Using SIP RTP CoS mark 5 Audio is at 16060 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably
2018 May 10
2
SIP Codec negotiation
I receive an INVITE/SDP containing: m=audio 11310 RTP/AVP 3 0 101 which I interpret as gsm, ulaw, rfc2833. and I reply with an OK/SDP containing: m=audio 15884 RTP/AVP 0 3 101 which I interpret as ulaw, gsm, rfc2833. How can I tell which codec was actually used for the call? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards
2006 Jan 27
0
pb with callerid
Since I passed from version 1.0 to the 1.2.3. I have Pb with the callerid. If somebody call with presentation of the number all is well. If somebody make call in masked number, i couldn't send a callerid to the phone. It is in a call center and i use the callerid to present the name of the number called to the operator. Before that went. To identify the sda, I use the assignment of the
2011 Apr 12
0
No subject
the legs separately as if they were not related to the same call. So the ingress leg negotiates ulaw, and despite it knowing that the peer also supports g729 fails the call since it's already decided on ulaw and the egress leg only accepts g729. If this is design intent I'm wondering if there is demand enough to justify a feature request? Any advice on how I can work around this issue?
2007 Feb 21
3
SIP 406 error - cause?
I'm working on calls coming in to an asterisk box as H.323, and going out as SIP to a remote device (a VoiceMaster). The remote device is refusing the calls with SIP error 406 (Not Acceptable). I have attached the SIP debug output below. It looks like codecs overlaps - can anyone see why the call is being refused? (Note that I'm not registering with the remote SIP device, just
2007 Nov 20
0
sl75 wlan not able of being pickuped?
Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being cancelled. No problems with other phones (polycom, grandstream). Attached the complete sip debug log
2007 Apr 26
1
Can asterisk record the duration of users putting 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
2008 Mar 10
0
Audiocodes MP124-FXS replying BUSY when line is not.
Hello, Has anybody seen that Audiocodes gateway is replying with "486 Busy here" when it's actually not (last call ended ~15 seconds ago). I see this quite often. From other logs i see that previous call ends at 11:13:01, then app_queue tries to dial at 11:13:14 and fails numerous times, before succeeding at 11:14:02 I have attached sample SIP debug log: Any ideas what i could
2003 Nov 06
2
this is the code that breaks outgoing calls on grandstream
Here is the diff from chan_sip.c 15 days ago and 16 days ago. 15 days ago is the point outgoing calls made via grandstream budgetone stopped working. Any help on why it breaks? Any possible fix? /tmp# diff asterisk/channels/chan_sip.c asterisk.works/channels/chan_sip.c 289d288 < int capability; 3921,3922d3919 < p->capability = user->capability;
2009 May 06
2
Understanding Codecs
Hi, I'm having problems with an asterisk server that's not offering Codecs for ulaw and alaw as it should. I've three servers in total: a1, a2 and "b" A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see
2009 Jan 08
0
SIP message routed back to mysql
Hello! * Version: 1.6.0.3-rc1 Scenario: * -> Proxy -> routed back to myself (The only thing changing is the Request URI) (And the Record-Route, Via that are added, of course). Outgoing Context is faxserver-out, incoming context is faxserver (at least should be). Outgoing context is straight forward: [faxserver-out] exten => _X.,1,NoOP(FAXOUT -- Connecting ${CALLERID(all)} ->
2008 Nov 07
1
Outgoing SIP calls dropped after 30 seconds.
Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Problem: Make a call on a Polycom 320 IP phone to
2004 Jan 14
1
Codec matching weirdness
I am experiencing a problem that from list archive it appears others are running into. When I dial from Cisco 7960 via the * to Free World Dialup destinations that supports G.729 the call fails. The major error from the debug log is Jan 15 00:11:14 NOTICE[22545]: channel.c:1481 ast_set_read_format: Unable to find a path from G729A to ULAW Jan 15 00:11:14 NOTICE[22545]: channel.c:1451
2007 Jan 10
1
caller id not transferred to SIP device
Hello, I'm wondering why asterisk is not transferring the callerid to the sip device. Scenario as follows: sangoma <---> zaptel <---> asterisk <---> sip <---> SIP-Device zaptel is reporting the callerid, but in the sip packages the sip-address shows unknown as user part, as this sip debug package shows: Executing Dial("Zap/62-1",
2007 Nov 29
0
[Copfilter] Copy of quarantined email - *** SPAM *** [7.4/6.0] Re: Asterisk <-> Nortel Phone Switch
[asterisk-users] Asterisk <-> Nortel Phone Switch Date: Thu, 29 Nov 2007 07:52:17 +0000 (GMT) X-Mailer: sendEmail-1.52 MIME-Version: 1.0 Content-Type: multipart/mixed; boundary="----MIME delimiter for sendEmail-20854.4017086787" This is a multi-part message in MIME format. To properly display this message you need a MIME-Version 1.0 compliant Email program. ------MIME delimiter