# chan_sip to PJSIP Migration: sip_to_pjsip.py, Option Mapping, Tests Source: https://srvscripts.com/guides/chan-sip-to-pjsip-migration/ Updated: 2026-10-06 Publisher: srvScripts (https://srvscripts.com/) **Short answer:** chan_sip was removed in Asterisk 21, so any `sip.conf` setup must move to `pjsip.conf` before you upgrade. Run the official converter (`contrib/scripts/sip_to_pjsip/sip_to_pjsip.py sip.conf pjsip.conf`) to get a first draft, then fix what it misses: `qualify`, inbound auth on trunks and anything in its “Non mapped elements” block. On FreePBX use `fwconsole convert2pjsip` and `fwconsole trunks --convert2pjsip` instead. We ran these commands on our lab server (Debian 12, FreePBX 17.0.33, Asterisk 22.11) on 6 October 2026: the upstream converter on a sample sip.conf, and the fwconsole help for FreePBX’s converters. The conversion output below is from that run; we did not convert a production system. ## Why you have to migrate chan_sip was deprecated for years and the Asterisk project removed it in Asterisk 21. The current LTS releases (Asterisk 22, and Asterisk 24 when it ships) have only `chan_pjsip`. Our lab’s Asterisk 22.11 lists `chan_pjsip.so` but no `chan_sip.so` at all: ``` asterisk -rx "module show like chan_" ``` FreePBX 17 ships with Asterisk 22 by default, so a FreePBX 16 system with chan_sip extensions or trunks hits the same wall when you move. Asterisk 20 LTS still has chan_sip, but it enters security-fix-only status on 19 October 2026 and reaches end of life on 19 October 2027. PJSIP is not a renamed chan_sip. One `[peer]` becomes several objects: | PJSIP object | Holds | chan_sip equivalent | | --- | --- | --- | | type=endpoint | Call settings: context, codecs, DTMF, NAT, media | Most peer options | | type=aor | Where to reach it: max contacts, static contact, qualify | host=, qualify= | | type=auth | Username and password | secret=, username= | | type=identify | Match inbound traffic by source IP | host= on IP-authenticated trunks | | type=registration | Outbound registration to a provider | register => line | | type=transport | Bind address, protocol, NAT addresses | [general] bind and NAT settings | ## Run sip_to_pjsip.py The script lives in the Asterisk source tree at `contrib/scripts/sip_to_pjsip/` and needs its two helper files (`astconfigparser.py`, `astdicts.py`) beside it. The Asterisk 22 version runs on Python 3. Fetch all three into a work folder, away from `/etc/asterisk`: ``` mkdir -p /root/pjsip-migration && cd /root/pjsip-migration for f in sip_to_pjsip.py astconfigparser.py astdicts.py; do curl -fsSO https://raw.githubusercontent.com/asterisk/asterisk/22/contrib/scripts/sip_to_pjsip/$f done cp -p /etc/asterisk/sip*.conf . python3 sip_to_pjsip.py --help ``` Help output on lab3: ``` Usage: sip_to_pjsip.py [options] [input-file [output-file]] Converts the chan_sip configuration input-file to the chan_pjsip output-file. The input-file defaults to 'sip.conf'. The output-file defaults to 'pjsip.conf'. Options: -h, --help show this help message and exit -p PREFIX, --prefix=PREFIX output prefix for include files -q, --quiet don't print messages to stdout ``` Run it from the folder that holds `sip.conf` and any files it `#include`s, and write to a new file so nothing live is touched: ``` python3 sip_to_pjsip.py sip.conf pjsip-converted.conf ``` FreePBX 17 also carries an older copy of this script under `/var/www/html/admin/modules/core/sip_to_pjsip/`. On our lab it is Python 2 code and fails under Debian 12’s Python 3.11 with `SyntaxError: Missing parentheses in call to 'print'`. Use the upstream copy above, or FreePBX’s own fwconsole converters. ## What the converter produced on our lab Our test `sip.conf` had a NAT setting in `[general]`, a registering trunk and one phone (all example values): ``` [general] udpbindaddr=0.0.0.0:5060 externip=203.0.113.10 localnet=192.168.1.0/255.255.255.0 register => bob-trunk:ExamplePass1@sip.example.com [bob-trunk] type=peer host=sip.example.com username=bob-trunk secret=ExamplePass1 fromuser=bob-trunk insecure=port,invite context=from-trunk disallow=all allow=ulaw,alaw dtmfmode=rfc2833 qualify=yes [201] type=friend host=dynamic secret=ExamplePass2 context=from-internal nat=force_rport,comedia qualify=yes directmedia=no disallow=all allow=ulaw,g722 dtmfmode=rfc2833 callerid="Bob" ``` The converter wrote this (exactly as produced, comment banner shortened): ``` ; Non mapped elements start [general] udpbindaddr = 0.0.0.0:5060 [bob-trunk] username = bob-trunk [201] qualify = yes ; Non mapped elements end [transport-udp] type = transport protocol = udp bind = 0.0.0.0:5060 external_media_address = 203.0.113.10 external_signaling_address = 203.0.113.10 local_net = 192.168.1.0/255.255.255.0 [reg_sip.example.com] type = registration retry_interval = 20 max_retries = 10 expiration = 120 transport = transport-udp outbound_auth = auth_reg_sip.example.com client_uri = sip:bob-trunk@sip.example.com server_uri = sip:sip.example.com [auth_reg_sip.example.com] type = auth password = ExamplePass1 username = bob-trunk [bob-trunk] type = aor contact = sip:bob-trunk@sip.example.com [bob-trunk] type = identify endpoint = bob-trunk match = sip.example.com [bob-trunk] type = auth username = bob-trunk password = ExamplePass1 [bob-trunk] type = endpoint context = from-trunk dtmf_mode = rfc4733 disallow = all allow = ulaw,alaw from_user = bob-trunk auth = bob-trunk outbound_auth = bob-trunk aors = bob-trunk [201] type = aor max_contacts = 1 [201] type = auth username = 201 password = ExamplePass2 [201] type = endpoint context = from-internal dtmf_mode = rfc4733 disallow = all allow = ulaw,g722 rtp_symmetric = yes force_rport = yes rewrite_contact = yes direct_media = no callerid = "Bob" auth = 201 outbound_auth = 201 aors = 201 ``` What to fix by hand before using it: - **qualify was not converted.** `qualify=yes` landed in the “Non mapped” block for the phone and vanished for the trunk. Add `qualify_frequency=60` (or your preferred interval in seconds) to each `type=aor`. - **The trunk got inbound auth.** `auth = bob-trunk` on the trunk endpoint means Asterisk will challenge calls from the provider. Providers usually do not answer challenges; with an `identify` match you normally keep only `outbound_auth`. The chan_sip `insecure=port,invite` line was dropped without a note. - **The phone got outbound_auth it does not need.** Harmless, but you can remove it. - **Registration and auth names changed** (`reg_sip.example.com`, `auth_reg_sip.example.com`). Update any monitoring that checks registration names. - **Everything in the “Non mapped elements” block** needs a manual decision. Delete the block once handled; it is not valid PJSIP. ## Option mapping for common chan_sip settings | chan_sip (sip.conf) | PJSIP (pjsip.conf) | Object | | --- | --- | --- | | secret=, username= | password=, username=, auth_type=userpass | auth | | host=dynamic | max_contacts=1 (or more) | aor | | host=203.0.113.10 | contact=sip:203.0.113.10:5060 plus an identify | aor, identify | | qualify=yes | qualify_frequency=60 | aor | | nat=force_rport,comedia (or nat=yes) | rtp_symmetric=yes, force_rport=yes, rewrite_contact=yes | endpoint | | directmedia=no | direct_media=no | endpoint | | dtmfmode=rfc2833 | dtmf_mode=rfc4733 | endpoint | | fromuser=, fromdomain= | from_user=, from_domain= | endpoint | | context=, allow=, disallow=, callerid= | same names | endpoint | | register => user:pass@host | type=registration + outbound_auth | registration, auth | | externip=, localnet= | external_media_address=, external_signaling_address=, local_net= | transport | | udpbindaddr= | bind= on a protocol=udp transport | transport | | insecure=invite | No direct option: match the trunk with identify and do not set inbound auth | endpoint, identify | The NAT mapping follows the Asterisk documentation: `nat=yes` becomes the three options above, `nat=route` becomes `force_rport=yes` and `rewrite_contact=yes` without symmetric RTP. Our [PJSIP NAT settings generator](/tools/pjsip-nat-generator/) writes the full NAT block for you. ## Migrating on FreePBX FreePBX stores extensions and trunks in its database and writes the `pjsip.*.conf` files itself, so do not paste converted files into `/etc/asterisk`. Use its tools. FreePBX 17.0.33 on our lab has: ``` fwconsole convert2pjsip --all # every chan_sip extension fwconsole convert2pjsip --range=5000-5100,6020 # selected extensions fwconsole trunks --list # see trunks and their technology fwconsole trunks --convert2pjsip=all # or a single trunk ID ``` When restoring a FreePBX 16 backup onto 17, `fwconsole backup` has matching switches: `--convertchansipexts2pjsip`, `--convertchansiptrunks2pjsip`, or `--skipchansipexts` and `--skipchansiptrunks` to leave them out. Our [FreePBX 16 to 17 upgrade guide](/guides/upgrade-freepbx-16-to-17/) covers that path. Converted phones must re-register to the PJSIP port. If PJSIP used a different port than chan_sip on the old system (FreePBX 13-16 often ran PJSIP on 5060 and chan_sip on 5160, or the other way round), update phone provisioning or the Bind Port in Asterisk SIP Settings before the cut-over. ## Test the migration Load the new configuration on a test box or during a maintenance window. Transport changes need a full restart, not a reload: ``` asterisk -rx "core restart when convenient" asterisk -rx "pjsip show transports" asterisk -rx "pjsip show registrations" asterisk -rx "pjsip show endpoints" asterisk -rx "pjsip show contacts" ``` Then work through this list: - Each provider registration shows **Registered**. - Each phone appears in `pjsip show contacts` with status **Avail** and a round-trip time. - Inbound call from the provider reaches the right context (watch with `pjsip set logger on`; a `401` or `403` back to the provider points to the inbound auth problem above). - Outbound call with correct caller ID. - Audio both ways on an external call and on a phone behind NAT. - DTMF in voicemail or an IVR (checks `dtmf_mode`). - Transfers, hold, call parking and BLF (hints use `PJSIP/201` instead of `SIP/201`). Search your custom dialplan for leftovers: `grep -rn "SIP/" /etc/asterisk/extensions*.conf`. Any `Dial(SIP/...)`, `SIPPEER()` or `SIP_HEADER()` must change to `PJSIP/...`, `PJSIP_ENDPOINT()`/`PJSIP_CONTACT()` and `PJSIP_HEADER()`. **Official documentation:** [Asterisk: Migrating from chan_sip to res_pjsip](https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Migrating-from-chan_sip-to-res_pjsip/) · [sip_to_pjsip source (Asterisk 22)](https://github.com/asterisk/asterisk/tree/22/contrib/scripts/sip_to_pjsip) · [Asterisk versions and EOL dates](https://docs.asterisk.org/About-the-Project/Asterisk-Versions/) **Related:** [PJSIP NAT Settings Generator: Asterisk, FreePBX, Issabel, VitalPBX, 3CX](/tools/pjsip-nat-generator/) · [PJSIP Behind NAT: Asterisk and FreePBX Settings for Two-Way Audio](/guides/pjsip-nat-asterisk-freepbx/) · [Install Asterisk 22 LTS on Debian 13: Step-by-Step Guide](/guides/install-asterisk-22-debian/) · [Troubleshoot SIP Calls with sngrep and the Asterisk PJSIP Logger](/guides/troubleshoot-sip-sngrep/) · [SIP Response Codes: Lookup for Every SIP Error, With Causes and Fixes](/tools/sip-response-codes/) **See also:** [sip.conf to pjsip.conf Converter for Asterisk](/tools/sip-to-pjsip-converter/) · [Asterisk CLI Commands Cheat Sheet: PJSIP, Calls, Dialplan, Logs](/guides/asterisk-cli-commands-cheat-sheet/) · [PJSIP Endpoint Unreachable or Unavailable: Fix Qualify and NAT](/guides/pjsip-endpoint-unreachable-qualify/) · [FreePBX PJSIP Trunk Setup: Registration, IP Auth, Routes, NAT](/guides/freepbx-pjsip-trunk-setup/) ## Frequently asked questions ### Which Asterisk version removed chan_sip? Asterisk 21. Asterisk 20 LTS is the last LTS release with chan_sip; it goes security-fix-only on 19 October 2026 and end of life on 19 October 2027. ### Does sip_to_pjsip.py convert everything? No. The Asterisk docs say it is not meant to cover every configuration. In our test it skipped qualify, dropped insecure, and added inbound auth to a trunk, so review the output line by line. ### Can I run chan_sip and PJSIP side by side during migration? Only on Asterisk 20 or older, and they must listen on different ports. On Asterisk 21 and later only PJSIP exists. ### How do I convert FreePBX chan_sip extensions? Run fwconsole convert2pjsip –all or with –range, and fwconsole trunks –convert2pjsip=all for trunks. Then fwconsole reload and re-register the phones. ### What replaces SIPPEER() and SIP_HEADER() in dialplan? Use PJSIP_ENDPOINT() or PJSIP_CONTACT() for peer data and PJSIP_HEADER() to read or add SIP headers.