similar to: Unable to lookup host in c= line,

Anyone else have issues with T.38 where the call drops after T.38 is attempted to be negotiated, with a message like the below? WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'

Hello everyone. I'm struggling to get T.38 faxing to work in Asterisk 1.6.1.13 with a SIP DID provider here in Brazil (GVT - Vox IP service). Here's my scenario: When faxes arrive by a specific DID, they are routed thru this simple macro: [macro-recebefax] exten => s,1,Set(DB(fax/count)=$[${DB(fax/count)} + 1]) exten => s,n,Set(FAXCOUNT=${DB(fax/count)}) exten =>

Hi all: I recently update a system from CVS (Asterisk CVS-09/25/03-15:58:42), and I receiving the following message: *CLI> WARNING[1187305408]: File chan_sip.c, Line 1864 (process_sdp): No compatible codecs! The "show codecs" command shows: *CLI> show codecs 1 (1 << 0) G.723.1 2 (1 << 1) GSM 4 (1 << 2) G.711 u-law 8 (1 << 3) G.711 A-law 16 (1 <<

When I get a SIP call, I get the following error: *CLI> NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is 'multipart/mixed;boundary="unique-boundary-1"', not 'application/sdp' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 5147771111, 1) == Spawn extension (incoming,

Hello, I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call. When a fax call is made Movistar send only T38 in the INVITE. Invite example: v=0 o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 s=sip call c=IN IP4 192.168.1.2 t=0 0 m=audio 6370 RTP/AVP 18 101 a=fmtp:18

Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk stop working after the upgrade. Here is the sip debug: --------------------------------------------------------------------------- <--- SIP read from 208.65.xxx.xxx:5060 ---> INVITE sip:1778xxxxxxx at 10.11.22.77:5060 SIP/2.0 Via: SIP/2.0/UDP 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-d9175178645e9146-1---d8754z-;rport Via:

Hi, I've upgraded my asterisk 1.4 to the version 1.8.11. After making some adjustments to the configuration files to port it to the new version, calls between registered phones in asterisk, work fine, but inbound calls coming from the SIP trunk I have with a telco to asterisk, don't work anymore. I don't know why!... This is the SDP portion that comes in the INVITE messages of calls

I'm trying to figure out how to handle our fax line when we switch to our asterisk for voice. After a lot of reading and poking about I have concluded, as have many others it would seem, that the best thing to do is either to have a separate pstn fax line or use some sort of internet faxing service rather than try and make faxing work in a way it's not meant to over voip lines.

Hi, I am recently updated my asterisk 1.4.1 to 1.6.2.6 because of fax service. I have installed spandsp 0.0.5, then 0.0.6 pre 17 as well because to have support. app_fax is available in menuselect and loaded into my asterisk. The upgrade went without any problems. ast*CLI> module show like fax Module Description Use Count app_fax.so

Hi All, I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6, But when making A Call from SIP Client, I got cli Warning ... and no call has been made. My Sip Client is using lib java peers client http://peers.sourceforge.net/ with standard codec PCMU/PCMA [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia

Retrying, falling of the list some how :-( -------- Original Message -------- Subject: feeling n00b again Date: 2018-08-20 09:51 From: asterisk at a-domani.nl To: asterisk-users at lists.digium.com Hi all, Long time ago, I followed a Asterisk training, and both at work and at home, was able to deploy Asterisk, make all sorts of internal call (hard/soft voip-phones, incoming/outgoing,

I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default

Asterisk 1.2 has no support of t.38 whatsoever, the call will drop before t.38 is ever utilised, not even pass-thru. 1.4 Adds support for T.38 pass through only and no other sort of faxing, the endpoint must support T.38 and you must send your call to a T.38 gateway and you must not use NAT anywhere in your network and you must enable re-invites which could cause CDRs not to reflect the true

Hi All, Excited to see mention of T.38 passthru with 1.4.0.... BUT it seems like "Emperor's New Clothes" from my perspective. As I understand it, T.38 passthru only works via a re-invite mechanism - so two T.38 devices try talking fax and request a T.38 media stream. Asterisk is now savvy enough to realise it can't do it, so opts out - "Hey Guys, I don't know

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

Does anyone have a good list of links to the various presentations at Astricon, specifically one including a link to the performance analysis by those guys from Belgium? I would love to get a closer look at their graphs because it was impossible to read them, and I was pretty close to the front! -- Kristian Kielhofner

While not strictly Asterisk related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html -- Kristian Kielhofner

Hi folks, In an effort to save bandwidth (our 7905s run over a WAN) we've switched from ulaw to g729a. We purchased 4 licenses from Digium (4 SIP clients, low call volume), and they seem to have been accepted: [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e == Found license

Hey, I am here at Astricon and about to go down to registration. Is there any interest in pictures if I take my digital camera? I am sure that someone is already doing this. (Probably someone official). I would take pictures of each day and upload them to my website if anyone is interested. Let me know! -- Kristian Kielhofner