similar to: RE: Pb DTMF with Asterisk vs Cirpack Transit, , Node

Salut Guy, I have the same problem with a Cirpack (B3G carrier) What I see is that you use sip info to detect DTMF. The problem is that there is no normalisation on the content of the sip info frame for dtmf detection. First, asterisk try to detect the header "application/dtmf-relay" and you have the header "application/dtmf" see line 6069 of /channels/chan_sip.c function

Hi all, I've a problem in DTMF dialog between * and a Cicpack Transit Node Class 4-5. The call initiat by a mgcp phone pass by the cirpack and arrive in SIP on *, everything is ok (negociation and phone call) but when we try to use the voicemail, Asterisk don't understand DTMF. Here are some logs (SIP debug on) on a DTMF '2' receive :

Hi, I've got an * box with 4 E1 (TE405P) and a Class 4 Cirpack switch (www.cirpack.com). <IP Network>--<*>--<Cirpack>--<Public PSTN Network> ISDN interco is EuroIsdn, ports are configured ccs, hdb3, crc4. Cirpack is Network, * is Terminal/User. As I encountered some pb with Sip to Zap transcoding (* to Cirpack way poor quality, the other way fine), I tryed to

Hi, has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto or more info about needed Asterisk SW and setup ? Thanks in advance, regards, Rob.

Hi list, After a recent upgrade to Asterisk v1.4.14, my message log is now filling up with the following error messages: <-------------> [Dec 29 17:24:52] WARNING[10655]: chan_sip.c:6645 determine_firstline_parts: Bad request protocol Packet --- (1 headers 0 lines) --- bitis*CLI> <--- SIP read from 82.101.62.99:5060 ---> Cirpack KeepAlive Packet <-------------> Seeing

We are running HEAD from last night and 1.0.5 and 1.0.3 and 1.0.2 and they all are running just fine in production environments each handling thousands of calls a day. I suppose reliability depends upon what you are using, but for our purposes they all are very stable. I could do without the memory leaks though. MATT--- -----Original Message----- From: Florian Lefeuvre

Hi, I've got an * box with 4 E1 (TE405P) and a Class 4 Cirpack switch (www.cirpack.com). <IP Network>--<*>-[TE405P]-<Cirpack>-<Public PSTN Network> ISDN interco is EuroIsdn, ports are configured ccs, hdb3, crc4. Cirpack is Network, * is Terminal/User. As I encountered some pb with Sip to Zap transcoding (* to Cirpack way poor quality, the other way fine), I

Hi, I'm trying to get asterisk to work with the Cirpack Softswitch. All I need for now is that asterisk should forward all calls to the Cirpack. My sip.conf files looks like: [general]

I was messing about integration of a Cirpack softswitch with Asterisk and banged my head against a problem previously noted on the list. http://lists.digium.com/pipermail/asterisk-users/2003-November/026436.ht ml What is the status of this problem? Has it been fixed? I scrambled through chan_sip.c, but couldn't find ay reference to "multipart". Regards, Jesper Dalberg

Hi all!! I have a really weird problem. I upgraded 2 Asterisk 1.2 boxes to Asterisk 1.4.18. Both are home PBX's and both boxes register to a SIP DID at exactly same provider. One box runs without errors on the console, the other box keeps repeating : [Feb 11 23:40:29] WARNING[11292]: chan_sip.c:6705 determine_firstline_parts: Bad request protocol Packet When i set debug on, it seems to

hi all after getting some (or - a lot) of messages from nagios, claiming asterisk to be down, I found this out... astping xxx times repeatedly, and the manager fails to start. this little script was used for testing. below, I've pasted the output from 'ps axfv' before and after the DoS, showing asterisk having allocated ~2GB RAM. roy #!/usr/bin/perl -w use strict; my $i = 0;

I'm trying to impute data using the mi package, but after running through almost the entire first round of imputations (which takes quite a while), it throws this error (I'll include the whole output prior to the error for context). Does anyone know what is causing it, or how I can fix it? More specifically, how can I tell what is throwing the error so I know what to fix? Is

Hi I request your help because i don't have actually a solution at my problems. I have a Asterisk Server in 1.6 Connected at a SIP Provider This provider supply me 2 numbers: 003318364xxxx (official number) 081169xxxx (Nddi Number) When i receive a call on the 081169xxxx, he don't use the extension. He use the 003318364xxxx extension. SIP Debug: <--- SIP read from

Some CVS upgrade in the last day or two has broken the recognition of DTMF eg in Voicemail. I'm running the latest CVS as of now. I'm getting the error... *CLI> -- Executing VoiceMailMain("SIP/phone1-e0dd", "") in new stack -- Playing 'vm-login' (language 'en') **Here I push a button** May 9 18:26:18 WARNING[98311]: chan_sip.c:5027

Doe anyone have Caller ID name working on incoming calls with the following setup: cisco 2611XM recieving calls on a PRI. cisco talks SIP to the asterisk box. Phones (cisco 7940s) talk SIP to asterisk. I currently have verified the cisco router is receiving caller ID name from the ILEC, and it appears to be sending it to the asterisk. But, it looks like it comes in after the call, and *

Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 100 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP

Hi, I'm getting this error when registering with SIP server using Asterisk 1.4.10 and Freepbx... I'm getting this error no matter what I try to setup in sip.conf : - I'm getting confused whether options are maxexpirey=36000 or maxexpiry=36000 ? - Can I solve this with some settings in sip.conf or is this problem harder ? - I've read something about Asterisk's bug on this

Good evening! I was wondering one thing, I'm using freepbx to configure my asterisk server and I have a problem with some inbound calls. When I receive a call to an INVITE sip:01xxxxxx at myip.com I an set an inbound route! It matches a DID number. How can I route an INVITE sip:s at myip.com? The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it

After a migration to asterisk 1.6, I don't receive sip incoming calls anymore. As fas as I understand the SIP debug traces, my server receives the request and reject it: ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ <--- SIP read from UDP:212.27.52.5:5060 ---> INVITE sip:s at 192.168.4.2:5060;transport=udp SIP/2.0 Call-ID: 25151-WW-0eaf098b-2f615ac60 at

Hello all, I've been using * for quite some time and yesterday I decided to add sipbroker to my config. It was pretty simple and it works for some numbers (e.g. I can call *258-9123, UK date & time - which is on the "phone numbers you can call" page -) but fails for some others. For example I've got a friend who's at freephonie so to call him, I would dial