similar to: T38 Passthrough 1.6.1.12-rc1 Good Results

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

Hello List, I have been trying to setup T38 gatewaying with the following setup SIP ->Asterisk -> DAHDI TE410P with Libss7 -> TELCO I'm using asterisk Asterisk 1.8.3.2 and DAHDI Version: SVN-trunk-r9697M Echo Canceller: HWEC I'm aware there's no support for T38 gateway but I have been trying to get the patches https://issues.asterisk.org/view.php?id=13405 to work. It seems

Hi All, I'm in the lab with Asterisk 1.6.1.12 and several ATA's testing T38. I hit a snag with the Grandstream HT502. It only seems to nail up a session at 9600bps. The Grandstream GXW4104 nails up consistently at 14400bps. I'm using the same equipment in the same configuration, just switching out the ATA. I have the latest firmware on each unit. Any ideas on what could cause

> Asterisk Project Security Advisory - ASA-2007-010 > > +------------------------------------------------------------------------+ > | Product | Asterisk | > |--------------------+---------------------------------------------------| > | Summary | Two stack buffer overflows in SIP

> Asterisk Project Security Advisory - ASA-2007-010 > > +------------------------------------------------------------------------+ > | Product | Asterisk | > |--------------------+---------------------------------------------------| > | Summary | Two stack buffer overflows in SIP

Hi, I'm trying to send faxes using Asterisk 1.4 and T38 with sip but Asterisk rejects the t38. Anybody know if is possible to transmit t38 fax with Asterisk 1.4? following settings: --- sip.conf --- [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw context=from-outside t38pt_udptl=yes [operator] qualify=no nat=yes host=189.160.126.201 dtmfmode=rfc2833

I found a bug in the latest T38 passthrough patches, the effect is that a non-SIP call after being put on hold is then lost, no resume is possible. The fix is to be applied in the chan_sip.c file: } else { /* No bridged peer with T38 enabled*/ transmit_response_with_sdp(p, "200 OK", req, 1); } - } + } else transmit_response_with_sdp(p, "200

Hi All, I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet !

We are testing the new T38 passthrough code (SVN-oej-t38passthrough-r13347): - we are using a Sipura SPA-2100 as the T.38 user device - we are using a Patton SmartNode 2400 as the T.38/PRI gateway - we are using Asterisk in the middle We have the following in the [general] section of our sip.conf: t38pt_udptl = yes t38pt_rtp = yes When a fax call comes in from the SmartNode to Asterisk

Hi all the people in the list! I'm new on this list, this is my first post. I configured asterisk 1.6 with freepbx 2.7 and dahdi to send faxes with t38modem conected to hylafax as a sip extension of asterisk. Everything is supposed to be configured fine, the faxes start sending, but at the middle of the transaction, it fails. The T.38 termination provider told me that they were receiving

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

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

Friends in the Asterisk.org community, There is a lot of cool stuff going on in Asterisk development, things that will change Asterisk and make it work better in your organisation, make it easier to sell in your area or give you more consulting oppurtunities - in short, functionality that will make a lot of sense for you users. However, developers can't really get anywhere without a

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

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

Hi Guys, Something I have noticed while dealing with T.38 and re-invites in Asterisk 1.4.22. I have a provider who re-invites with the following sdp (message flow PROVIDER_EQPMT -> ASTERISK): """ . v=0. o=SIP_5F9 123456 654322 IN IP4 CONN_IP_PROVIDER. s=-. c=IN IP4 CONN_IP_PROVIDER. t=0 0. m=audio 0 RTP/AVP 0. m=image 26858 udptl t38. a=T38FaxMaxBuffer:288.

Hello, I need help to solve a problem that I am having using Asterisk 13, PJSIP and T38. My setup is as follows: SIP Provider --> Asterisk 13 --> Patton --> Physical Fax I need to get the fax directly in T38 to Patton. The provider sends me the fax in T38. If I receive the T38 fax on Asterisk (using an hylafax device), I can properly receive the fax. If I send a T38 fax with Asterisk

Hi all, I have a question regarding using T38 for fax sending and here is my scenario: fax -> SIP ATA (T38 enabled) -> Asterisk #1 -> IAX TRUNK -> Asterisk #2 -> SIP ATA (T38 enabled) -> fax My question is, how can I know if I'm really using T38? is T38 information coming to the other side (because of SIP to IAX conversion) or just plain g711a data? I'm using Linksys

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

Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38