similar to: lastest spandsp-0.03pre1 don't compile

Displaying 20 results from an estimated 700 matches similar to: "lastest spandsp-0.03pre1 don't compile"

2006 Apr 26
0
Re: [Serusers] Sip t38 gateway tests
Thanks for these informations I would have prefer to receive them from asterisk-users instead of serusers !! May be they are sleeping . Ok i have not installed spandsp because of i don't find some scripts like in hylafax for mail2fax fax2mail i've just patched chan_sip.c Regards Harry --- Alexandr Dubovikov <shurik@start4.info> a ?crit : > On Wed, Apr 26, 2006 at 11:27:01AM
2008 Sep 12
4
Custom build kernel patch fails big time.
I am trying to build a custom kernel for the HIPL code (infrahip.hip.fi), using the patch for the Linux 2.6.18 kernel. I followed all the instructions for getting the kernel source and making a custom kernel provided on the wiki. The rpmbuild fails with the following messages: Get an error on this step (figured out my other problem): Patch #40000
2003 Feb 23
0
Question about some Cisco-specific code in "rtp.c"
The file "rtp.c" currently contains the following hack for special-case handling of some Cisco-specific protocol: } else if (payloadtype == 121) { /* CISCO proprietary DTMF bridge */ f = process_type121(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); As I noted in my earlier message, I'm
2010 Jun 22
1
UDPTL T38 via NAT
Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-----[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from
2009 Dec 10
1
Asterisk 1.6.1.11 Fax
Hello, We're trying to receive faxes on the Asterisk server, but for the time being T.38 negotiation fails. The SDP that the Asterisk reINVITE sends contains these lines: ---------------------- m=image 4968 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400
2010 Jan 29
1
callerid not working over sip
Calling from my home using Asterisk 1.6.2.1 to an office extension (Asterisk 1.6.1.13) the callerid is not honored: Home: -- Starting simple switch on 'DAHDI/1-1' -- Executing [170 at internal:1] Answer("DAHDI/1-1", "") in new stack -- Executing [170 at internal:2] NoOp("DAHDI/1-1", "Context: office-extensions") in new stack
2008 Feb 15
1
DialPlan help with Analog Fax Machine
I'm struggling to get my dialplan to work with a simple analog fax machine. I have TDM400B zaptel card with an FXO and FXS port. I have the FXO port connected to the POTS machine and the FAX machine connected to the FXS port. The FAX machine itself works fine, I can FAX outgoing messages fine. I can also dial the FAX extension from the internal context, the FAX machine answers and I
2010 Feb 20
1
Fax, T38 and NAT
Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38
2008 Aug 11
1
Intermittent T.38 pass through
Hi All, I've been testing reliability with t.38 faxing pass through with * 1.4.21.1, Linksys ATA's 2102 x 2, Sharp UX-B800SE and Cannon ImageClass D880. cannon> <2102 #1> <SIP> <*> <SIP> <2102 #2> <sharp Started out with default settings on all devices, configured Asterisk to handle T.38 pass through, the configuration I believe is solid. I get
2017 Jun 16
3
asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'
On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote: <snip> > > t38modem and asterisk are using > > m=image 35622 udptl t38 > ^^^^^ > > Provider uses > > m=image 35622 UDPTL t38 > ^^^^^ > > Could this be a problem? If I'm sending internal only, it's always > lowercase. Looking at the tests we have we
2012 Feb 02
1
T38 faxing - UDPTL creation failed
Hello guys. When I am trying to send fax through T38 to linksys SPA (properly configured etc. - I have tried it with other systems), I'm getting error and fax is not delivered. I'm getting this errors in asterisk.log: WARNING[687] udptl.c: No UDPTL ports remaining ERROR[687] chan_sip.c: UDPTL creation failed WARNING[687] udptl.c: No UDPTL ports remaining then, couple lines down:
2009 Jul 03
0
e164.org and tollfree ENUM records
Recently, I've been having issues with the URIs returned from e.164.org and toll free calls. It seems that the URIs that are returned from ENUMQUERY and ENUMRESULT are no longer the proper numbering schemes that the poviders use. I've been using the following [enum] template in my outbound route for quite some time with great success until recently. [enum](!) exten =>
2012 Apr 27
1
No UDPTL ports remaining
Hi all, Lately, I've been seeing more and more instances where I get a flood of warning messages like this: [Apr 26 14:09:50] WARNING[21054] udptl.c: No UDPTL ports remaining The next thing I know, my server is dropping calls and starting to misbehave. I use fax via T.38, so I can't just turn udptl off. I could expand the port range, but I suspect that will just mask the situation.
2017 Jun 18
2
asterisk 13.16. - sigseg during negotiation
Hello! unchanged asterisk crashes during udptl / t.38 negotiation with telekom - they do not support t.38 / udptl. In detail: fax client -> asterisk -> telekom -> easybell -> asterisk -> fax server Fax server sends t.38 reinvite via asterisk to easybell. Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 2447581897 4 IN IP4 46.17.15.23
2009 Nov 22
1
transferring SIP call: no voice
I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk B. Both are behind NAT, but port forwarded. I get the connection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax. CLI on A looks right: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 ==
2023 Apr 28
1
Asterisk translates 200 OK + SDP into 488 not acceptable here after both side agreed on codec.
Hi List Asterisk 16.28.0 in use. PJSIP in use Two endpoints Both using IPv6 One Endpoint on UDP, the other via TLS. Both with: t38_udptl=yes ;fax_detect=yes ;fax_detect_timeout=30 rtp_ipv6=yes Both sides are T.38 capable and detect fax tone so no need for fax detection on asterisk. Voice calls between the two work fine. But on a Fax call, I see this situation: A <=> Asterisk
2013 Jan 04
0
T38MaxBitRate issue on fax passthrough
Having an issue with receiving faxes, but when I pass through the fax. Currently, I receive the fax with Digium's Fax for Asterisk, store it and the initiate an outbound call to our fax server. (XMedius Fax). This works, but we would prefer to have Asterisk simply route the call directly to the fax server and take the store and forward out of the equation. When I do that, however, the
2013 Jan 15
4
Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable
Hi, I configured Asterisk 10 for inbound fax, for couple of weeks I didn't see any issues until today. The setup I configured for inbound fax is quite simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38 protocol and later Asterisk stores/forwards the fax to specific end user. The configuration I made in sip.conf for enabling T38 is listed below; t38pt_udptl =
2016 Feb 04
0
AST-2016-003: Remote crash vulnerability when receiving UDPTL FAX data.
Asterisk Project Security Advisory - AST-2016-003 Product Asterisk Summary Remote crash vulnerability when receiving UDPTL FAX data. Nature of Advisory Denial of Service Susceptibility Remote
2007 Jul 12
0
No subject
reliable, 40+ faxes with 5 different fax machines tested, plus an efax service. The problem is the other way, PRI inbound to fax, the call doesn't setup between the GW and Asterisk. I have the issued narrowed down to the SIP messaging between the DMG and Asterisk. The DMG invite sends to asterisk: m=audio 49016 RTP/AVP 0 101 [notice the m=audio] a=rtpmap:0 PCMU/8000/1 a=rtpmap:101